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Supervised Transfer Not working


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I have the following set up

 

CellPhone <-> Hypermedia Gateway <-> PBXnSIP <-> Microsoft SpeechServer

 

I am trying to implement the supervised transfer. The Incoming call works fine. I hear the IVR Prompt. Speech server makes an outbound call (consultation call) to another phone. This works Ok too I hear the IVR Prompt and enter my response. However, after the I enter the response on teh consultation call, and the transfer occurs, I do not hear anything on either phones. after sometime the outbound call hangs up on the phone. However, PBXnSIP still shows teh call as active.

 

I had problems with TCP Timeout during outbound call because speech server did not authenticate and call timed out after 8 secs and PBXnSIP support team helped by providing the latest .exe files that solved the tcp timeout issue. So the Outbound leg is working fine. Now I just need to transfer the call and I will complete this project (http://doctoroncalljamaica.com) . Please see the logs below. I again see TCP Timeout in the logs...

 

 

 

INVITE sip:8768776075@192.168.1.13 SIP/2.0

Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-1d4212783c10c531-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.12:5060>

To: <sip:8768776075@192.168.1.13>

From: <sip:0012018881440@192.168.1.12:5060>;tag=b853b310

Call-ID: OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub, 100rel, em

User-Agent: HG4000/1.0

Content-Length: 345

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4000 RTP/AVP 4 18 18 18 18 0 8 101

a=rtpmap:4 G723/8000

a=rtpmap:18 G729/8000

a=rtpmap:18 G729a/8000

a=rtpmap:18 G729b/8000

a=rtpmap:18 G729ab/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110615020610: SIP Tx tcp:192.168.1.12:9224:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-1d4212783c10c531-1---d8754z-;rport=9224

From: <sip:0012018881440@192.168.1.12:5060>;tag=b853b310

To: <sip:8768776075@192.168.1.13>;tag=b90bf2c68a

Call-ID: OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

CSeq: 1 INVITE

Content-Length: 0

 

 

[5] 20110615020610: Using <sip:0012018881440@192.168.1.12:5060;user=phone> as redirect source address

[5] 20110615020610: SIP Tx tcp:192.168.1.13:6060:

INVITE sip:8768776075@192.168.1.13:6060;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:52543;branch=z9hG4bK-885b2b907e6916aef67ac0402280350b;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

To: <sip:8768776075@192.168.1.13:6060;user=phone>

Call-ID: 7226e2d7@pbx

CSeq: 16835 INVITE

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:52543;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

Related-Call-ID: OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Type: application/sdp

Content-Length: 327

 

v=0

o=- 32718 32718 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 10920 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110615020610: SIP Rx tcp:192.168.1.13:6060:

SIP/2.0 100 Trying

FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

TO: <sip:8768776075@192.168.1.13:6060;user=phone>

CSEQ: 16835 INVITE

CALL-ID: 7226e2d7@pbx

VIA: SIP/2.0/TCP 192.168.1.13:52543;branch=z9hG4bK-885b2b907e6916aef67ac0402280350b;rport

CONTENT-LENGTH: 0

 

 

[5] 20110615020610: SIP Tx tcp:192.168.1.12:9224:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-1d4212783c10c531-1---d8754z-;rport=9224

From: <sip:0012018881440@192.168.1.12:5060>;tag=b853b310

To: <sip:8768776075@192.168.1.13>;tag=b90bf2c68a

Call-ID: OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

CSeq: 1 INVITE

Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 251

 

v=0

o=- 1422 1422 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 50780 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110615020610: SIP Rx tcp:192.168.1.13:6060:

SIP/2.0 302 Moved Temporarily

FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

TO: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=9d5d30897e

CSEQ: 16835 INVITE

CALL-ID: 7226e2d7@pbx

VIA: SIP/2.0/TCP 192.168.1.13:52543;branch=z9hG4bK-885b2b907e6916aef67ac0402280350b;rport

CONTACT: <sip:8768776075@192.168.1.13:52479;user=phone;transport=Tcp;maddr=192.168.1.13;x-mss-call-id=7226e2d7%40pbx>

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[7] 20110615020610: Call 7226e2d7@pbx: Clear last INVITE

[5] 20110615020610: SIP Tx tcp:192.168.1.13:6060:

ACK sip:8768776075@192.168.1.13:6060;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:52543;branch=z9hG4bK-885b2b907e6916aef67ac0402280350b;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=9d5d30897e

Call-ID: 7226e2d7@pbx

CSeq: 16835 ACK

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:52543;transport=tcp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020610: Redirecting call

[5] 20110615020610: SIP Tx tcp:192.168.1.13:52479:

INVITE sip:8768776075@192.168.1.13:52479;user=phone;transport=Tcp;maddr=192.168.1.13;x-mss-call-id=7226e2d7%40pbx SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-9947fb33db4d71b414e25acf3deba21d;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

To: <sip:8768776075@192.168.1.13:6060;user=phone>

Call-ID: 7226e2d7@pbx

CSeq: 16836 INVITE

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:52544;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

Related-Call-ID: OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Type: application/sdp

Content-Length: 327

 

v=0

o=- 32718 32718 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 10920 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110615020610: SIP Rx tcp:192.168.1.13:52479:

SIP/2.0 100 Trying

FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

TO: <sip:8768776075@192.168.1.13:6060;user=phone>

CSEQ: 16836 INVITE

CALL-ID: 7226e2d7@pbx

VIA: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-9947fb33db4d71b414e25acf3deba21d;rport

CONTENT-LENGTH: 0

 

 

[5] 20110615020610: SIP Rx tcp:192.168.1.13:52479:

SIP/2.0 180 Ringing

FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

TO: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=8E0ADA2D20;tag=5b60ff999

CSEQ: 16836 INVITE

CALL-ID: 7226e2d7@pbx

VIA: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-9947fb33db4d71b414e25acf3deba21d;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[5] 20110615020610: SIP Rx tcp:192.168.1.13:52479:

SIP/2.0 200 OK

FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

TO: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=8E0ADA2D20;tag=5b60ff999

CSEQ: 16836 INVITE

CALL-ID: 7226e2d7@pbx

VIA: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-9947fb33db4d71b414e25acf3deba21d;rport

CONTACT: <sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13>;automata

CONTENT-LENGTH: 194

CONTENT-TYPE: application/sdp

ALLOW: UPDATE

SERVER: RTCC/3.0.0.0

ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

 

v=0

o=- 0 0 IN IP4 192.168.1.13

s=Microsoft Speech Server session

c=IN IP4 192.168.1.13

t=0 0

m=audio 13440 RTP/AVP 0 8 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

 

[7] 20110615020610: Call 7226e2d7@pbx: Clear last INVITE

[5] 20110615020610: SIP Tx tcp:192.168.1.13:52479:

ACK sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13 SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-cc94dd24be9fd5e8c08ff1830236a068;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=5b60ff999;epid=8E0ADA2D20

Call-ID: 7226e2d7@pbx

CSeq: 16836 ACK

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:52544;transport=tcp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020610: SIP Tx tcp:192.168.1.12:9224:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-1d4212783c10c531-1---d8754z-;rport=9224

From: <sip:0012018881440@192.168.1.12:5060>;tag=b853b310

To: <sip:8768776075@192.168.1.13>;tag=b90bf2c68a

Call-ID: OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

CSeq: 1 INVITE

Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Content-Type: application/sdp

Content-Length: 251

 

v=0

o=- 1422 1422 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 50780 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110615020612: SIP Tx tcp:192.168.1.12:9224:

BYE sip:0012018881440@192.168.1.12:5060 SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:5060;branch=z9hG4bK-ba981f3f1fe00badc3844a63077c2c5d;rport

From: <sip:8768776075@192.168.1.13>;tag=b90bf2c68a

To: <sip:0012018881440@192.168.1.12:5060>;tag=b853b310

Call-ID: OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

CSeq: 21381 BYE

Max-Forwards: 70

Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

Content-Length: 0

 

 

[5] 20110615020612: SIP Tx tcp:192.168.1.13:52479:

BYE sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13 SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-9d38c8c5d436718a6eb0a76aec6f17a9;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=5b60ff999

Call-ID: 7226e2d7@pbx

CSeq: 16837 BYE

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:52544;transport=tcp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020612: SIP Rx tcp:192.168.1.13:52479:

SIP/2.0 200 OK

FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=33746

TO: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=5b60ff999;epid=8E0ADA2D20

CSEQ: 16837 BYE

CALL-ID: 7226e2d7@pbx

VIA: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-9d38c8c5d436718a6eb0a76aec6f17a9;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[7] 20110615020612: Call 7226e2d7@pbx: Clear last request

[5] 20110615020612: BYE Response: Terminate 7226e2d7@pbx

[5] 20110615020612: SIP Rx tcp:192.168.1.12:9224:

SIP/2.0 481 Call/Transaction Does Not Exist

Via: SIP/2.0/TCP 192.168.1.13:5060;branch=z9hG4bK-ba981f3f1fe00badc3844a63077c2c5d;rport=5060

To: <sip:0012018881440@192.168.1.12:5060>;tag=b853b310

From: <sip:8768776075@192.168.1.13>;tag=b90bf2c68a

Call-ID: OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

CSeq: 21381 BYE

Accept-Language: en

Content-Length: 0

 

 

[7] 20110615020612: Call OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.: Clear last request

[5] 20110615020612: BYE Response: Terminate OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

[5] 20110615020705: SIP Rx tcp:192.168.1.12:9224:

INVITE sip:8768776075@192.168.1.13 SIP/2.0

Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-b5ca0f04f101ea54-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.12:5060>

To: <sip:8768776075@192.168.1.13>

From: <sip:0012018881440@192.168.1.12:5060>;tag=97716957

Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub, 100rel, em

User-Agent: HG4000/1.0

Content-Length: 345

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4000 RTP/AVP 4 18 18 18 18 0 8 101

a=rtpmap:4 G723/8000

a=rtpmap:18 G729/8000

a=rtpmap:18 G729a/8000

a=rtpmap:18 G729b/8000

a=rtpmap:18 G729ab/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110615020705: SIP Tx tcp:192.168.1.12:9224:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-b5ca0f04f101ea54-1---d8754z-;rport=9224

From: <sip:0012018881440@192.168.1.12:5060>;tag=97716957

To: <sip:8768776075@192.168.1.13>;tag=2af52cac74

Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

CSeq: 1 INVITE

Content-Length: 0

 

 

[5] 20110615020705: Using <sip:0012018881440@192.168.1.12:5060;user=phone> as redirect source address

[5] 20110615020705: SIP Tx tcp:192.168.1.13:6060:

INVITE sip:8768776075@192.168.1.13:6060;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:52543;branch=z9hG4bK-63b9477d8a581c99013ba28baf356258;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

To: <sip:8768776075@192.168.1.13:6060;user=phone>

Call-ID: 6e790856@pbx

CSeq: 858 INVITE

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:52543;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

Related-Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Type: application/sdp

Content-Length: 327

 

v=0

o=- 23513 23513 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 30302 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110615020705: SIP Rx tcp:192.168.1.13:6060:

SIP/2.0 100 Trying

FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

TO: <sip:8768776075@192.168.1.13:6060;user=phone>

CSEQ: 858 INVITE

CALL-ID: 6e790856@pbx

VIA: SIP/2.0/TCP 192.168.1.13:52543;branch=z9hG4bK-63b9477d8a581c99013ba28baf356258;rport

CONTENT-LENGTH: 0

 

 

[5] 20110615020705: SIP Tx tcp:192.168.1.12:9224:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-b5ca0f04f101ea54-1---d8754z-;rport=9224

From: <sip:0012018881440@192.168.1.12:5060>;tag=97716957

To: <sip:8768776075@192.168.1.13>;tag=2af52cac74

Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

CSeq: 1 INVITE

Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 251

 

v=0

o=- 3356 3356 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 52784 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110615020705: SIP Rx tcp:192.168.1.13:6060:

SIP/2.0 302 Moved Temporarily

FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

TO: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=49bbd880fd

CSEQ: 858 INVITE

CALL-ID: 6e790856@pbx

VIA: SIP/2.0/TCP 192.168.1.13:52543;branch=z9hG4bK-63b9477d8a581c99013ba28baf356258;rport

CONTACT: <sip:8768776075@192.168.1.13:52479;user=phone;transport=Tcp;maddr=192.168.1.13;x-mss-call-id=6e790856%40pbx>

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[7] 20110615020705: Call 6e790856@pbx: Clear last INVITE

[5] 20110615020705: SIP Tx tcp:192.168.1.13:6060:

ACK sip:8768776075@192.168.1.13:6060;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:52543;branch=z9hG4bK-63b9477d8a581c99013ba28baf356258;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=49bbd880fd

Call-ID: 6e790856@pbx

CSeq: 858 ACK

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:52543;transport=tcp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020705: Redirecting call

[5] 20110615020705: SIP Tx tcp:192.168.1.13:52479:

INVITE sip:8768776075@192.168.1.13:52479;user=phone;transport=Tcp;maddr=192.168.1.13;x-mss-call-id=6e790856%40pbx SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-140e26839d6c2ca053d84501bf7bb611;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

To: <sip:8768776075@192.168.1.13:6060;user=phone>

Call-ID: 6e790856@pbx

CSeq: 859 INVITE

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:52544;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

Related-Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Type: application/sdp

Content-Length: 327

 

v=0

o=- 23513 23513 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 30302 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110615020705: SIP Rx tcp:192.168.1.13:52479:

SIP/2.0 100 Trying

FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

TO: <sip:8768776075@192.168.1.13:6060;user=phone>

CSEQ: 859 INVITE

CALL-ID: 6e790856@pbx

VIA: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-140e26839d6c2ca053d84501bf7bb611;rport

CONTENT-LENGTH: 0

 

 

[5] 20110615020705: SIP Rx tcp:192.168.1.13:52479:

SIP/2.0 180 Ringing

FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

TO: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=8E0ADA2D20;tag=649944c4f

CSEQ: 859 INVITE

CALL-ID: 6e790856@pbx

VIA: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-140e26839d6c2ca053d84501bf7bb611;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[5] 20110615020705: SIP Rx tcp:192.168.1.13:52479:

SIP/2.0 200 OK

FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

TO: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=8E0ADA2D20;tag=649944c4f

CSEQ: 859 INVITE

CALL-ID: 6e790856@pbx

VIA: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-140e26839d6c2ca053d84501bf7bb611;rport

CONTACT: <sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13>;automata

CONTENT-LENGTH: 194

CONTENT-TYPE: application/sdp

ALLOW: UPDATE

SERVER: RTCC/3.0.0.0

ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

 

v=0

o=- 0 0 IN IP4 192.168.1.13

s=Microsoft Speech Server session

c=IN IP4 192.168.1.13

t=0 0

m=audio 35840 RTP/AVP 0 8 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

 

[7] 20110615020705: Call 6e790856@pbx: Clear last INVITE

[5] 20110615020705: SIP Tx tcp:192.168.1.13:52479:

ACK sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13 SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-8dad98c585973e9425c80587931750c9;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=649944c4f;epid=8E0ADA2D20

Call-ID: 6e790856@pbx

CSeq: 859 ACK

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:52544;transport=tcp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020705: SIP Tx tcp:192.168.1.12:9224:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-b5ca0f04f101ea54-1---d8754z-;rport=9224

From: <sip:0012018881440@192.168.1.12:5060>;tag=97716957

To: <sip:8768776075@192.168.1.13>;tag=2af52cac74

Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

CSeq: 1 INVITE

Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Content-Type: application/sdp

Content-Length: 251

 

v=0

o=- 3356 3356 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 52784 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110615020705: SIP Rx tcp:192.168.1.12:9224:

ACK sip:8768776075@192.168.1.13:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-7c1251351bd6f75e-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.12:5060>

To: <sip:8768776075@192.168.1.13>;tag=2af52cac74

From: <sip:0012018881440@192.168.1.12:5060>;tag=97716957

Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

CSeq: 1 ACK

User-Agent: HG4000/1.0

Content-Length: 0

 

 

[5] 20110615020737: SIP Rx tcp:192.168.1.13:52479:

INVITE sip:0012018881440@192.168.1.13:52544;transport=tcp SIP/2.0

FROM: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=8E0ADA2D20;tag=649944c4f

TO: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

CSEQ: 1 INVITE

CALL-ID: 6e790856@pbx

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.1.13:52479;branch=z9hG4bK1b7861a6

CONTACT: <sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13;ms-opaque=5cf12f79b09db613>;automata

CONTENT-LENGTH: 206

USER-AGENT: RTCC/3.0.0.0

CONTENT-TYPE: application/sdp

 

v=0

o=- 0 0 IN IP4 192.168.1.13

s=Microsoft Speech Server session

c=IN IP4 192.168.1.13

t=0 0

m=audio 35840 RTP/AVP 0 8 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendonly

a=ptime:20

 

[5] 20110615020737: SIP Tx tcp:192.168.1.13:52479:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.1.13:52479;branch=z9hG4bK1b7861a6

From: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=649944c4f;epid=8E0ADA2D20

To: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

Call-ID: 6e790856@pbx

CSeq: 1 INVITE

Contact: <sip:0012018881440@192.168.1.13:52544;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Content-Type: application/sdp

Content-Length: 265

 

v=0

o=- 23513 23513 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 30302 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=recvonly

 

[5] 20110615020737: SIP Rx tcp:192.168.1.13:52479:

ACK sip:0012018881440@192.168.1.13:52544;transport=tcp SIP/2.0

FROM: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=8E0ADA2D20;tag=649944c4f

TO: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

CSEQ: 1 ACK

CALL-ID: 6e790856@pbx

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.1.13:52479;branch=z9hG4bK4ac3478c

CONTENT-LENGTH: 0

USER-AGENT: RTCC/3.0.0.0

 

 

[5] 20110615020737: SIP Rx tcp:192.168.1.13:52556:

INVITE sip:12012181444@192.168.1.13:5060;transport=tcp SIP/2.0

FROM: <sip:0012018881440@192.168.1.12:5060>;epid=8E0ADA2D20;tag=5454b672e0

TO: <sip:12012181444@192.168.1.13:5060;transport=tcp>

CSEQ: 2 INVITE

CALL-ID: 4d74728f-46c9-4a48-9268-c6b892a4ecc8

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.1.13:52556;branch=z9hG4bKcd18343

CONTACT: <sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13;ms-opaque=5cf12f79b09db613>;automata

CONTENT-LENGTH: 336

USER-AGENT: RTCC/3.0.0.0

CONTENT-TYPE: application/sdp

ALLOW: UPDATE

ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

 

v=0

o=- 0 0 IN IP4 192.168.1.13

s=Microsoft Speech Server session

c=IN IP4 192.168.1.13

t=0 0

m=audio 13440 RTP/AVP 114 115 4 0 8 97 101

a=rtpmap:114 x-msrta/16000

a=fmtp:114 bitrate=29000

a=rtpmap:115 x-msrta/8000

a=fmtp:115 bitrate=11800

a=rtpmap:97 RED/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

 

[5] 20110615020737: SIP Tx tcp:192.168.1.13:52556:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 192.168.1.13:52556;branch=z9hG4bKcd18343

From: <sip:0012018881440@192.168.1.12:5060>;tag=5454b672e0;epid=8E0ADA2D20

To: <sip:12012181444@192.168.1.13:5060;transport=tcp>;tag=9936c19f3f

Call-ID: 4d74728f-46c9-4a48-9268-c6b892a4ecc8

CSeq: 2 INVITE

Content-Length: 0

 

 

[5] 20110615020737: Using <sip:0012018881440@192.168.1.12:5060;user=phone> as redirect source address

[5] 20110615020737: SIP Tx tcp:192.168.1.12:5060:

INVITE sip:12012181444@192.168.1.12;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

To: <sip:12012181444@192.168.1.12;user=phone>

Call-ID: 1df6b336@pbx

CSeq: 19159 INVITE

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:52557;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Diversion: <tel:45>;reason=unconditional;screen=no;privacy=off

Related-Call-ID: 4d74728f-46c9-4a48-9268-c6b892a4ecc8

P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

Content-Type: application/sdp

Content-Length: 327

 

v=0

o=- 28916 28916 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 38056 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110615020737: SIP Tx tcp:192.168.1.13:52556:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TCP 192.168.1.13:52556;branch=z9hG4bKcd18343

From: <sip:0012018881440@192.168.1.12:5060>;tag=5454b672e0;epid=8E0ADA2D20

To: <sip:12012181444@192.168.1.13:5060;transport=tcp>;tag=9936c19f3f

Call-ID: 4d74728f-46c9-4a48-9268-c6b892a4ecc8

CSeq: 2 INVITE

Contact: <sip:12012181444@192.168.1.13:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Content-Type: application/sdp

Content-Length: 263

 

v=0

o=- 6605 6605 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 10552 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110615020737: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

Call-ID: 1df6b336@pbx

CSeq: 19159 INVITE

Content-Type: application/sdp

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110615020742: Did not receive ACK, disconnecting call OWNjNWEyMDJhYTY5ZDcwYWU2NGVlMTQyNGM3MjkxZTA.

[5] 20110615020750: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

Call-ID: 1df6b336@pbx

CSeq: 19159 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[7] 20110615020750: Call 1df6b336@pbx: Clear last INVITE

[5] 20110615020750: SIP Tx udp:192.168.1.12:5060:

ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

Call-ID: 1df6b336@pbx

CSeq: 19159 ACK

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020750: SIP Tx tcp:192.168.1.13:52556:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.1.13:52556;branch=z9hG4bKcd18343

From: <sip:0012018881440@192.168.1.12:5060>;tag=5454b672e0;epid=8E0ADA2D20

To: <sip:12012181444@192.168.1.13:5060;transport=tcp>;tag=9936c19f3f

Call-ID: 4d74728f-46c9-4a48-9268-c6b892a4ecc8

CSeq: 2 INVITE

Contact: <sip:12012181444@192.168.1.13:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Content-Type: application/sdp

Content-Length: 263

 

v=0

o=- 6605 6605 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 10552 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110615020750: SIP Rx tcp:192.168.1.13:52556:

ACK sip:12012181444@192.168.1.13:5060;transport=tcp SIP/2.0

FROM: <sip:0012018881440@192.168.1.12:5060>;epid=8E0ADA2D20;tag=5454b672e0

TO: <sip:12012181444@192.168.1.13:5060;transport=tcp>;tag=9936c19f3f

CSEQ: 2 ACK

CALL-ID: 4d74728f-46c9-4a48-9268-c6b892a4ecc8

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.1.13:52556;branch=z9hG4bKd3a69aa1

CONTENT-LENGTH: 0

USER-AGENT: RTCC/3.0.0.0

 

 

[5] 20110615020750: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

Call-ID: 1df6b336@pbx

CSeq: 19159 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110615020750: SIP Tx udp:192.168.1.12:5060:

ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

Call-ID: 1df6b336@pbx

CSeq: 19159 ACK

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020751: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

Call-ID: 1df6b336@pbx

CSeq: 19159 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110615020751: SIP Tx udp:192.168.1.12:5060:

ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

Call-ID: 1df6b336@pbx

CSeq: 19159 ACK

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020753: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

Call-ID: 1df6b336@pbx

CSeq: 19159 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110615020753: SIP Tx udp:192.168.1.12:5060:

ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

Call-ID: 1df6b336@pbx

CSeq: 19159 ACK

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020757: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

Call-ID: 1df6b336@pbx

CSeq: 19159 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110615020757: SIP Tx udp:192.168.1.12:5060:

ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

Call-ID: 1df6b336@pbx

CSeq: 19159 ACK

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020801: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

Call-ID: 1df6b336@pbx

CSeq: 19159 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110615020801: SIP Tx udp:192.168.1.12:5060:

ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

Call-ID: 1df6b336@pbx

CSeq: 19159 ACK

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

Content-Length: 0

 

 

[6] 20110615020805: SIP TCP/TLS timeout on 192.168.1.13:6060, closing connection

[5] 20110615020805: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

Call-ID: 1df6b336@pbx

CSeq: 19159 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110615020805: SIP Tx udp:192.168.1.12:5060:

ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

Call-ID: 1df6b336@pbx

CSeq: 19159 ACK

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020809: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

Call-ID: 1df6b336@pbx

CSeq: 19159 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110615020809: SIP Tx udp:192.168.1.12:5060:

ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

Call-ID: 1df6b336@pbx

CSeq: 19159 ACK

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020813: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

Call-ID: 1df6b336@pbx

CSeq: 19159 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110615020813: SIP Tx udp:192.168.1.12:5060:

ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

Call-ID: 1df6b336@pbx

CSeq: 19159 ACK

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020817: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

Call-ID: 1df6b336@pbx

CSeq: 19159 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110615020817: SIP Tx udp:192.168.1.12:5060:

ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

Call-ID: 1df6b336@pbx

CSeq: 19159 ACK

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020821: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557

Contact: <sip:12012181444@192.168.1.12:5060;user=phone>

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

Call-ID: 1df6b336@pbx

CSeq: 19159 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110615020821: SIP Tx udp:192.168.1.12:5060:

ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

Call-ID: 1df6b336@pbx

CSeq: 19159 ACK

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020828: SIP Rx tcp:192.168.1.12:9224:

BYE sip:8768776075@192.168.1.13:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-6c29ed1dac6bad0e-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.12:5060>

To: <sip:8768776075@192.168.1.13>;tag=2af52cac74

From: <sip:0012018881440@192.168.1.12:5060>;tag=97716957

Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

CSeq: 2 BYE

User-Agent: HG4000/1.0

Reason: SIP;description="ACK not received"

Content-Length: 0

 

 

[5] 20110615020828: SIP Tx tcp:192.168.1.12:9224:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-6c29ed1dac6bad0e-1---d8754z-;rport=9224;received=192.168.1.12

From: <sip:0012018881440@192.168.1.12:5060>;tag=97716957

To: <sip:8768776075@192.168.1.13>;tag=2af52cac74

Call-ID: ZjI2ZmQwNjAwYmNlMzhhY2FlNTJhYTQxNWRmODMxN2Y.

CSeq: 2 BYE

Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

User-Agent: snom-PBX/2011-4.2.1.4009

Content-Length: 0

 

 

[5] 20110615020828: SIP Tx tcp:192.168.1.13:52479:

BYE sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13 SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-8fee055997a827e58c90fb6c7e9671e2;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=649944c4f

Call-ID: 6e790856@pbx

CSeq: 860 BYE

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:52544;transport=tcp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020828: SIP Rx tcp:192.168.1.13:52479:

SIP/2.0 200 OK

FROM: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=130

TO: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=649944c4f;epid=8E0ADA2D20

CSEQ: 860 BYE

CALL-ID: 6e790856@pbx

VIA: SIP/2.0/TCP 192.168.1.13:52544;branch=z9hG4bK-8fee055997a827e58c90fb6c7e9671e2;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[7] 20110615020828: Call 6e790856@pbx: Clear last request

[5] 20110615020828: BYE Response: Terminate 6e790856@pbx

[5] 20110615020837: SIP Rx udp:192.168.1.13:41032:

SUBSCRIBE sip:1000@192.168.1.13 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:41032;branch=z9hG4bK-d8754z-b34c6224c862d35c-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:1000@192.168.1.13:41032>

To: "1000"<sip:1000@192.168.1.13>

From: "1000"<sip:1000@192.168.1.13>;tag=161e10eb

Call-ID: OGQxMWQwZmFmYjQwMjUxZWU4M2NmNzkyNDQ5ZjgwZmQ.

CSeq: 1 SUBSCRIBE

Expires: 300

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

User-Agent: X-Lite 4 release 4.0 stamp 58832

Event: message-summary

Content-Length: 0

 

 

[5] 20110615020837: SIP Tx udp:192.168.1.13:41032:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.1.13:41032;branch=z9hG4bK-d8754z-b34c6224c862d35c-1---d8754z-;rport=41032

From: "1000" <sip:1000@192.168.1.13>;tag=161e10eb

To: "1000" <sip:1000@192.168.1.13>;tag=e2eaf47150

Call-ID: OGQxMWQwZmFmYjQwMjUxZWU4M2NmNzkyNDQ5ZjgwZmQ.

CSeq: 1 SUBSCRIBE

Content-Length: 0

 

 

[5] 20110615020913: SIP Tx tcp:192.168.1.13:52556:

BYE sip:CommServer.creditfree.local:52479;transport=Tcp;maddr=192.168.1.13;ms-opaque=5cf12f79b09db613 SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:5060;branch=z9hG4bK-fec7a8bfdfa5cbe6c65c284973f55bc3;rport

From: <sip:12012181444@192.168.1.13:5060;transport=tcp>;tag=9936c19f3f

To: <sip:0012018881440@192.168.1.12:5060>;tag=5454b672e0;epid=8E0ADA2D20

Call-ID: 4d74728f-46c9-4a48-9268-c6b892a4ecc8

CSeq: 31135 BYE

Max-Forwards: 70

Contact: <sip:12012181444@192.168.1.13:5060;transport=tcp>

Content-Length: 0

 

 

[5] 20110615020913: SIP Tx udp:192.168.1.12:5060:

BYE sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-c31f3d66d57f59090580890239bf9e84;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

Call-ID: 1df6b336@pbx

CSeq: 19160 BYE

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020913: SIP Rx tcp:192.168.1.13:52556:

SIP/2.0 200 OK

FROM: <sip:12012181444@192.168.1.13:5060;transport=tcp>;tag=9936c19f3f

TO: <sip:0012018881440@192.168.1.12:5060>;tag=5454b672e0;epid=8E0ADA2D20

CSEQ: 31135 BYE

CALL-ID: 4d74728f-46c9-4a48-9268-c6b892a4ecc8

VIA: SIP/2.0/TCP 192.168.1.13:5060;branch=z9hG4bK-fec7a8bfdfa5cbe6c65c284973f55bc3;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[7] 20110615020913: Call 4d74728f-46c9-4a48-9268-c6b892a4ecc8: Clear last request

[5] 20110615020913: BYE Response: Terminate 4d74728f-46c9-4a48-9268-c6b892a4ecc8

[5] 20110615020913: SIP Tr udp:192.168.1.12:5060:

BYE sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-c31f3d66d57f59090580890239bf9e84;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

Call-ID: 1df6b336@pbx

CSeq: 19160 BYE

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020921: Last message repeated 4 times

[6] 20110615020921: SIP TCP/TLS timeout on 192.168.1.12:5060, closing connection

[5] 20110615020924: SIP Tr udp:192.168.1.12:5060:

BYE sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-c31f3d66d57f59090580890239bf9e84;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

Call-ID: 1df6b336@pbx

CSeq: 19160 BYE

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

Content-Length: 0

 

 

[6] 20110615020928: SIP TCP/TLS timeout on 192.168.1.13:52479, closing connection

[5] 20110615020928: SIP Tr udp:192.168.1.12:5060:

BYE sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-c31f3d66d57f59090580890239bf9e84;rport

From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005

To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131

Call-ID: 1df6b336@pbx

CSeq: 19160 BYE

Max-Forwards: 70

Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110615020945: Last message repeated 5 times

[7] 20110615020945: Call 1df6b336@pbx: Clear last request

[5] 20110615020945: BYE Response: Terminate 1df6b336@pbx

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Looking at the log, the below messages are being repeated many times. So it looks like the ACK sent by PBX is never received by HG4000/1.0 (or somehow it is dropping it). Can you verify why HG4000/1.0 is re-transmitting the 200 OK message even after the ACK was sent by PBX?

 

[5] 20110615020821: SIP Rx tcp:192.168.1.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557
Contact: <sip:12012181444@192.168.1.12:5060;user=phone>
To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131
From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005
Call-ID: 1df6b336@pbx
CSeq: 19159 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER
Content-Type: application/sdp
Supported: replaces, norefersub
User-Agent: HG4000/1.0
Content-Length: 189

v=0
o=HG4000 0 0 IN IP4 192.168.1.12
s=HG4000-Session
c=IN IP4 192.168.1.12
t=0 0
m=audio 4010 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

[5] 20110615020821: SIP Tx udp:192.168.1.12:5060:
ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport
From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005
To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131
Call-ID: 1df6b336@pbx
CSeq: 19159 ACK
Max-Forwards: 70
Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>
P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>
Content-Length: 0

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Looking at the log, the below messages are being repeated many times. So it looks like the ACK sent by PBX is never received by HG4000/1.0 (or somehow it is dropping it). Can you verify why HG4000/1.0 is re-transmitting the 200 OK message even after the ACK was sent by PBX?

 

[5] 20110615020821: SIP Rx tcp:192.168.1.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.13:52557;branch=z9hG4bK-08322140e9b4e0e6ee3277b235cf955f;rport=52557
Contact: <sip:12012181444@192.168.1.12:5060;user=phone>
To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131
From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005
Call-ID: 1df6b336@pbx
CSeq: 19159 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER
Content-Type: application/sdp
Supported: replaces, norefersub
User-Agent: HG4000/1.0
Content-Length: 189

v=0
o=HG4000 0 0 IN IP4 192.168.1.12
s=HG4000-Session
c=IN IP4 192.168.1.12
t=0 0
m=audio 4010 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

[5] 20110615020821: SIP Tx udp:192.168.1.12:5060:
ACK sip:12012181444@192.168.1.12:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-9e70d3b58c580ffc8526933219ca6651;rport
From: <sip:0012018881440@192.168.1.12:5060;user=phone>;tag=34005
To: <sip:12012181444@192.168.1.12;user=phone>;tag=46648131
Call-ID: 1df6b336@pbx
CSeq: 19159 ACK
Max-Forwards: 70
Contact: <sip:0012018881440@192.168.1.13:5060;transport=udp>
[b]P-Asserted-Identity: "Forty Five" <sip:45@pbx.company.com;user=phone>[/b]Content-Length: 0

 

Am I missing something? there are no comments here....

FYI

I also changed the "Assume Calls from" in the Speech server trunk to 42 to match the "Asuume call comes from" in the Hypermedia Trunk. I hoped that it might help if teh two legs of the call originate from same extension. But it still does not work.

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Am I missing something? there are no comments here....

FYI

I also changed the "Assume Calls from" in the Speech server trunk to 42 to match the "Asuume call comes from" in the Hypermedia Trunk. I hoped that it might help if teh two legs of the call originate from same extension. But it still does not work.

 

Sorry. I will check teh HG 4000 Logs and let you know.

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You might be a "victim" of the 8 seconds TCP disconnect issue when the TCP/TLS connection did not register. if you send a pivate message to pbx_support and indicate what OS you have, your problem might go away already.

 

 

I had problems with TCP Timeout during outbound call because speech server did not authenticate and call timed out after 8 secs and PBXnSIP support team helped by providing the latest .exe files that solved the tcp timeout issue. So the Outbound leg is working fine. Now I just need to transfer the call and I will complete this project (http://doctoroncalljamaica.com) . Please see the logs below.

 

However, I again see TCP Timeout in the logs...

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Does the HG4000 support UDP? This is because the 200 Ok comes on TCP transport layer and contains a contact that is implies UDP. So the PBX sends it on UDP. Because the gateway repeats the 200 Ok, it seems that the ACK does not make it and this transport layer problem could be the reason.

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Does the HG4000 support UDP? This is because the 200 Ok comes on TCP transport layer and contains a contact that is implies UDP. So the PBX sends it on UDP. Because the gateway repeats the 200 Ok, it seems that the ACK does not make it and this transport layer problem could be the reason.

 

yes HG 4000 Supports both UDP and TCP. There is an option to set the protocol. If I set it to UDP, the incoming call in the first leg does not work. PBX just drops it. I dont even see it in the logs (even if I specify UDP on the trunk as transport). The moment I change the transport to TCP on HG 4000 , PBX sees the call and it works.

 

Is there any way We can force PBX to send TCP back to HG 4000?

 

Thanks,

 

Sanjeev.

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yes HG 4000 Supports both UDP and TCP. There is an option to set the protocol. If I set it to UDP, the incoming call in the first leg does not work. PBX just drops it. I dont even see it in the logs (even if I specify UDP on the trunk as transport). The moment I change the transport to TCP on HG 4000 , PBX sees the call and it works.

 

Is there any way We can force PBX to send TCP back to HG 4000?

 

Thanks,

 

Sanjeev.

 

I again tried to set HG4000 Transport = UDP. I also changed the Hypermedia Trunk in PBX to

sip:192.168.1.12:5060;transport=udp

 

The Call does not connect when I call. It look slike HG4000 i strying to make a call. Please see HG4000 Log below. But I see nothing in PBX (192.68.1.13). Should I configure something differetnly in PBX for UDP to work? I will also post a log of HG 4000 where the incoming and outgoing call are working but transfer doe not take place.

 

HG 4000 Log when Transport = UDP

 

 

[17/06-16:54:52.820] [debug] MakeNetCall (/452) source: 0012012181444 destination: 18768776075

[17/06-16:54:52.821] [debug] MakeNetCall (/452) best matching prefix for 18768776075 is 18768776075

[17/06-16:54:52.821] [info] (/452) Making call to: 8768776075@192.168.1.13

[17/06-16:54:52.822] [debug] GetLocalInfo (/452) GetLocalInfo received RTPAddr:192.168.0.3 wRTPPort:4000

[17/06-16:54:52.822] [debug] SetState (/452) 1

[17/06-16:54:52.823] [notice] call from 0012012181444 to 18768776075 dialing

[17/06-16:54:52.824] [debug] makeCall MakeCall from: 0012012181444 to: 8768776075@192.168.1.13

[17/06-16:54:52.825] [debug] ChangeContactAddress SetDefaultFrom report IP: 192.168.1.12

[17/06-16:54:52.825] [debug] createSdpContents FindMediaReportIP(192.168.1.13)=192.168.1.12

[17/06-16:54:52.826] [debug] createSdpContents addCodec(G723,8000)

[17/06-16:54:52.826] [debug] createSdpContents addCodec(G729,8000)

[17/06-16:54:52.827] [debug] createSdpContents addCodec(G729a,8000)

[17/06-16:54:52.827] [debug] createSdpContents addCodec(G729b,8000)

[17/06-16:54:52.827] [debug] createSdpContents addCodec(G729ab,8000)

[17/06-16:54:52.831] [debug] createSdpContents addCodec(PCMU,8000)

[17/06-16:54:52.832] [debug] createSdpContents addCodec(PCMA,8000)

[17/06-16:54:52.833] [debug] createSdpContents addCodec(telephone-event,8000)

[17/06-16:54:52.834] [debug] RegisterTokenForSession 1:0 -> UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452

[17/06-16:54:52.835] [debug] SetState Set session state to:eNotConnected

[17/06-16:54:52.836] [debug] makeCall Via:

[17/06-16:54:52.837] [debug] makeCall Via: 192.168.1.12

[17/06-16:54:52.838] [debug] makeCall From = [sip:0012012181444@192.168.1.12:5060]

[17/06-16:54:52.839] [debug] makeCall Sending INVITE [uAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ.]

[17/06-16:54:54.462] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

[17/06-16:54:54.463] Sending: [MGWResStatus /I124/AMG], Client ID:12

[17/06-16:54:54.463] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

[17/06-16:54:54.467] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

[17/06-16:54:54.468] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

[17/06-16:54:54.469] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

[17/06-16:54:54.471] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

[17/06-16:54:54.472] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

[17/06-16:54:54.473] HMCServer.signalSendChannels: sending to specific client

[17/06-16:54:56.460] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

[17/06-16:54:56.461] Sending: [MGWResStatus /I124/AMG], Client ID:12

[17/06-16:54:56.461] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

[17/06-16:54:56.465] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

[17/06-16:54:56.466] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

[17/06-16:54:56.467] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

[17/06-16:54:56.469] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

[17/06-16:54:56.470] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

[17/06-16:54:56.471] HMCServer.signalSendChannels: sending to specific client

[17/06-16:54:58.461] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

[17/06-16:54:58.462] Sending: [MGWResStatus /I124/AMG], Client ID:12

[17/06-16:54:58.462] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

[17/06-16:54:58.466] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

[17/06-16:54:58.467] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

[17/06-16:54:58.468] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

[17/06-16:54:58.470] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

[17/06-16:54:58.471] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

[17/06-16:54:58.472] HMCServer.signalSendChannels: sending to specific client

[17/06-16:54:58.891] * Received Packet: GenericReply /#90/@2b/x0,1/I2912/G

[17/06-16:55:00.471] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

[17/06-16:55:00.471] Sending: [MGWResStatus /I124/AMG], Client ID:12

[17/06-16:55:00.472] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

[17/06-16:55:00.476] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

[17/06-16:55:00.477] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

[17/06-16:55:00.478] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

[17/06-16:55:00.479] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

[17/06-16:55:00.481] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

[17/06-16:55:00.482] HMCServer.signalSendChannels: sending to specific client

[17/06-16:55:02.462] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

[17/06-16:55:02.463] Sending: [MGWResStatus /I124/AMG], Client ID:12

[17/06-16:55:02.463] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

[17/06-16:55:02.467] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

[17/06-16:55:02.468] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

[17/06-16:55:02.469] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

[17/06-16:55:02.472] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

[17/06-16:55:02.473] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

[17/06-16:55:02.475] HMCServer.signalSendChannels: sending to specific client

[17/06-16:55:04.460] HMCServer received [MGWResStatus /I124/AMG/S] from 192.168.1.14:62261

[17/06-16:55:04.461] Sending: [MGWResStatus /I124/AMG/S], Client ID:12

[17/06-16:55:04.462] MGWConnThread sending: [MGWResStatus /I124/AMG/S/#12]

[17/06-16:55:04.466] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/S/#12]

[17/06-16:55:04.467] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/S/#12

[17/06-16:55:04.468] Received from MGW: [MGWResStatusReply /I124/s0|0,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

[17/06-16:55:04.470] m_server.sendToNetwork:MGWResStatusReply /I124/s0|0,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

[17/06-16:55:04.471] Adding message [MGWResStatusReply /I124/s0|0,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n,n|n/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

[17/06-16:55:04.473] HMCServer.signalSendChannels: sending to specific client

[17/06-16:55:06.463] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

[17/06-16:55:06.464] Sending: [MGWResStatus /I124/AMG], Client ID:12

[17/06-16:55:06.465] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

[17/06-16:55:06.474] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

[17/06-16:55:06.475] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

[17/06-16:55:06.475] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

[17/06-16:55:06.477] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

[17/06-16:55:06.478] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

[17/06-16:55:06.479] HMCServer.signalSendChannels: sending to specific client

[17/06-16:55:08.459] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

[17/06-16:55:08.460] Sending: [MGWResStatus /I124/AMG], Client ID:12

[17/06-16:55:08.460] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

[17/06-16:55:08.464] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

[17/06-16:55:08.492] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

[17/06-16:55:08.493] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

[17/06-16:55:08.494] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

[17/06-16:55:08.495] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

[17/06-16:55:08.496] HMCServer.signalSendChannels: sending to specific client

[17/06-16:55:10.462] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

[17/06-16:55:10.463] Sending: [MGWResStatus /I124/AMG], Client ID:12

[17/06-16:55:10.464] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

[17/06-16:55:10.468] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

[17/06-16:55:10.469] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

[17/06-16:55:10.469] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

[17/06-16:55:10.470] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

[17/06-16:55:10.471] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

[17/06-16:55:10.472] HMCServer.signalSendChannels: sending to specific client

[17/06-16:55:12.461] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

[17/06-16:55:12.462] Sending: [MGWResStatus /I124/AMG], Client ID:12

[17/06-16:55:12.462] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

[17/06-16:55:12.466] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

[17/06-16:55:12.467] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

[17/06-16:55:12.467] Received from MGW: [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected/#12]

[17/06-16:55:12.468] m_server.sendToNetwork:MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected client:12 app:0

[17/06-16:55:12.469] Adding message [MGWResStatusReply /I124/g1|1|0012012181444|8768776075||s,i,i,i,i,i,f,f,f,f,f,f,f,f,f,f,f,f,f,f,i,i,i,i,f,f,f,f,f,f,f,f/RDisconnected] to eventQueue for Client 12 Application ID 0

[17/06-16:55:12.471] HMCServer.signalSendChannels: sending to specific client

[17/06-16:55:14.051] * Received Packet: HangingUp /A21/I452/o0/R10

[17/06-16:55:14.052] updateReplyContext: no effect

[17/06-16:55:14.053] Application:VoIP

[17/06-16:55:14.054] ID2App removing ID:452

[17/06-16:55:14.055] MGWConnThread sending: [HangingUp /A21/I452/o0/R10/#0]

[17/06-16:55:14.059] [debug] ProcessLine received from hgs: [HangingUp /A21/I452/o0/R10/#0]

[17/06-16:55:14.061] [debug] ProcessLine invoking HangingUp with tid: 452 cid: 0 params: /A21/I452/o0/R10/#0

[17/06-16:55:14.061] [debug] OnPhoneHangup (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452) media was not started - unregistering

[17/06-16:55:14.062] [debug] UnregisterSession 1:0 -> UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452

[17/06-16:55:14.063] [debug] Close (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452)

[17/06-16:55:14.063] [debug] Close (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452) closing session timeslot:0 handle: -1

[17/06-16:55:14.064] [debug] CloseResources (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452)

[17/06-16:55:14.064] [debug] CloseAudioChannel (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452) channel timeslot: 0 already closed

[17/06-16:55:14.065] [debug] CloseResources (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452) remove media resoruce: 14.0

[17/06-16:55:14.065] [debug] DisconnectEndpoints (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452)

[17/06-16:55:14.066] [debug] ClearNetCall (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452) Cause Code: 10 Converted Cause Code: 10

[17/06-16:55:14.066] [debug] UpdateStatsAndCDR (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452)

[17/06-16:55:14.067] [info] VoIP CDR: 1,2011-06-17 T 16:54:52,2011-06-17 T 16:54:52,0012012181444,8768776075,,2011-06-17 T 16:55:14,,,,,1,0,1,1,0,0,0,10

[17/06-16:55:14.068] [debug] ~tMGWSession (UAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ./452)

[17/06-16:55:14.068] [debug] clearCall ClearCall [uAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ.]

[17/06-16:55:14.069] [debug] setHangupReason m_nHangupQ931: 10 m_nHangupSIP: 480

[17/06-16:55:14.069] [debug] removeSession Call Id [uAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ.]: m_tokenToSession.erase

[17/06-16:55:14.070] [debug] ~tSessionInfo Call id [uAC:ODViOGJhM2EwNzljM2UzNGM5OTNhYWI1YzllZTMyNjQ.]: Calling ~tSessionInfo

[17/06-16:55:14.512] HMCServer received [MGWResStatus /I124/AMG] from 192.168.1.14:62261

[17/06-16:55:14.513] Sending: [MGWResStatus /I124/AMG], Client ID:12

[17/06-16:55:14.514] MGWConnThread sending: [MGWResStatus /I124/AMG/#12]

[17/06-16:55:14.518] [debug] ProcessLine received from hgs: [MGWResStatus /I124/AMG/#12]

[17/06-16:55:14.519] [debug] ProcessLine invoking MGWResStatus with tid: 124 cid: 12 params: /I124/AMG/#12

[17/06-16:55:14.520] Received

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This the LOG for HG4000 when Transport=TCP. Please note that Call connects. I Do an IVR and Do a supervised transfer. The outgoing call connects too. I do an IVR and authenticate the called party. Then I do a Transfer. I cannot hear anything on either phones.

 

HG 4000 when Transport = TCP

 

[17/06-17:24:59.113] [debug] exProceedEvent AudioCodes event: EV_ENHANCED_BIT_STATUS

[17/06-17:24:59.114] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:01.694] HMCServer received [ping] from 127.0.0.1:40150

[17/06-17:25:01.695] processRequestLines: reply: "pong " for client: 13

[17/06-17:25:04.243] * Received Packet: GenericReply /#90/@2b/x0,1/I3032/G

[17/06-17:25:05.503] * Received Packet: Dialing /A21/I456/o0/H5/S1/h14/s0/n18768776075/N0012012181444/i1

[17/06-17:25:05.505] updateReplyContext: no effect

[17/06-17:25:05.506] packStr=/A21/I456/o0/H5/S1/h14/s0/n18768776075/N0012012181444/i1

[17/06-17:25:05.507] strHW=5

[17/06-17:25:05.507] strTS=1

[17/06-17:25:05.508] No filters required for in address 21.1

[17/06-17:25:05.509] getApplication for:21.1-VoIP

[17/06-17:25:05.533] strCard=[21], nClientID=0

[17/06-17:25:05.534] To VoIP Trigger:Dialing /A21/I456/o0/H5/S1/h14/s0/n18768776075/N0012012181444/i1

[17/06-17:25:05.535] MGWConnThread sending: [Dialing /A21/I456/o0/H5/S1/h14/s0/n18768776075/N0012012181444/i1/#0]

[17/06-17:25:05.541] [debug] ProcessLine received from hgs: [Dialing /A21/I456/o0/H5/S1/h14/s0/n18768776075/N0012012181444/i1/#0]

[17/06-17:25:05.554] [debug] ProcessLine invoking Dialing with tid: 456 cid: 0 params: /A21/I456/o0/H5/S1/h14/s0/n18768776075/N0012012181444/i1/#0

[17/06-17:25:05.555] [debug] CreateOutgoingSession CID: 0 TID: 456 cardAddres: 21 direction: 1 dstHW: 14 dstTS: 0 srcHW: 5 srcTS: 1

[17/06-17:25:05.555] [debug] Init (/) start session active timer

[17/06-17:25:05.556] [debug] ReserveLocalMediaResources (/456) m_MediaResourcesInUse Set 14.0

[17/06-17:25:05.556] [debug] ReserveLocalMediaResources (/456) : LocalHwyTS 14:0 RemoteHwyTS 5:1

[17/06-17:25:05.556] [debug] RegisterSession 1:0 -> /456

[17/06-17:25:05.557] Received from MGW: [DialAck /A21/I456/x0,0/o1/#0]

[17/06-17:25:05.558] Application:VoIP

[17/06-17:25:05.559] Sending: [DialAck /A21/I456/x0,0/o1], Client ID:0

[17/06-17:25:05.559] Real session ID [456]

[17/06-17:25:05.602] [debug] MakeNetCall (/456) source: 0012012181444 destination: 18768776075

[17/06-17:25:05.604] [debug] MakeNetCall (/456) best matching prefix for 18768776075 is 18768776075

[17/06-17:25:05.605] [info] (/456) Making call to: 8768776075@192.168.1.13

[17/06-17:25:05.605] [debug] GetLocalInfo (/456) GetLocalInfo received RTPAddr:192.168.0.3 wRTPPort:4000

[17/06-17:25:05.606] [debug] SetState (/456) 1

[17/06-17:25:05.606] [notice] call from 0012012181444 to 18768776075 dialing

[17/06-17:25:05.607] [debug] makeCall MakeCall from: 0012012181444 to: 8768776075@192.168.1.13

[17/06-17:25:05.608] [debug] ChangeContactAddress SetDefaultFrom report IP: 192.168.1.12

[17/06-17:25:05.609] [debug] createSdpContents FindMediaReportIP(192.168.1.13)=192.168.1.12

[17/06-17:25:05.609] [debug] createSdpContents addCodec(G723,8000)

[17/06-17:25:05.610] [debug] createSdpContents addCodec(G729,8000)

[17/06-17:25:05.610] [debug] createSdpContents addCodec(G729a,8000)

[17/06-17:25:05.611] [debug] createSdpContents addCodec(G729b,8000)

[17/06-17:25:05.611] [debug] createSdpContents addCodec(G729ab,8000)

[17/06-17:25:05.612] [debug] createSdpContents addCodec(PCMU,8000)

[17/06-17:25:05.619] [debug] createSdpContents addCodec(PCMA,8000)

[17/06-17:25:05.619] [debug] createSdpContents addCodec(telephone-event,8000)

[17/06-17:25:05.620] [debug] RegisterTokenForSession 1:0 -> UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456

[17/06-17:25:05.620] [debug] SetState Set session state to:eNotConnected

[17/06-17:25:05.621] [debug] makeCall Via:

[17/06-17:25:05.621] [debug] makeCall Via: 192.168.1.12

[17/06-17:25:05.622] [debug] makeCall From = [sip:0012012181444@192.168.1.12:5060]

[17/06-17:25:05.624] [debug] makeCall Sending INVITE [uAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.]

[17/06-17:25:05.625] [debug] onTrying UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

[17/06-17:25:05.626] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:25:05.627] [debug] onNewSession UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.remoteIP: 192.168.1.13

[17/06-17:25:05.628] [debug] GetLocalInfo (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) GetLocalInfo received RTPAddr:192.168.0.3 wRTPPort:4000

[17/06-17:25:05.629] Received from MGW: [sysAlerting /A21/I456/x0,0/o1/#0]

[17/06-17:25:05.631] Application:VoIP

[17/06-17:25:05.632] Sending: [sysAlerting /A21/I456/x0,0/o1], Client ID:0

[17/06-17:25:05.633] Real session ID [456]

[17/06-17:25:05.683] [debug] OnCreateExternalRTPHandler created external rtp handler board: 192.168.0.3 rtp port: 4000 for token: UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

[17/06-17:25:05.683] [debug] onProvisional UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

[17/06-17:25:05.684] [debug] onEarlyMedia UAC - Starting media.

[17/06-17:25:05.685] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:25:05.686] [debug] updateRemoteMediaAddr Updated remote media address to: 192.168.1.13

[17/06-17:25:05.687] [debug] OnStartExternalRTPHandler start external rtp handler for token: UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

[17/06-17:25:05.688] [debug] mediaAllocResources Creating channel with payload 0

[17/06-17:25:05.689] [debug] setChannelparam setting channel parameters board: 0 bus: 1 timeslot: 0 payload: 0 mute DTMF: 0

[17/06-17:25:05.690] [debug] setChannelparam using SIP/RFC2833, payload: 101

[17/06-17:25:05.691] [debug] exOpenChannel Setting mapping: Channel=0 -> HW:TS=1.0

[17/06-17:25:05.693] [debug] exOpenChannel Created channel 0

[17/06-17:25:05.694] [debug] mediaAllocResources OK mediaAllocResources Bus#1 Bus#0 PayLoad=0

[17/06-17:25:05.695] [debug] ~stopwatch mediaAllocResources: 690 usec

[17/06-17:25:05.696] [debug] ~stopwatch openChannel: 742 usec

[17/06-17:25:05.697] [debug] CreateChannel (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) created channel timeslot: 0 handle: 0

[17/06-17:25:05.698] [debug] StartMediaStream (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) localMediaAddr:192.168.0.3 localMediaPort:4000 remoteMediaAddr:192.168.1.13 remoteMediaPort:53830 Payload:0

[17/06-17:25:05.699] [debug] mediaActivateRTP_RTCPChannel acActivateRTP_RTCPChannel( IPPrec=0, nTOS=0, tx=0,rx=0,ChannelHandle 0 ) returned 0

[17/06-17:25:05.700] [debug] ~stopwatch mediaActivateRTP_RTCPChannel: 276 usec

[17/06-17:25:05.701] [debug] startMedia Started media, accepting call [uAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.]

[17/06-17:25:05.714] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:05.725] [debug] onReadyToSend UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

[17/06-17:25:05.726] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:25:05.726] [debug] onAnswer UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

[17/06-17:25:05.727] Received from MGW: [Answering /A21/I456/x0,0/o1/#0]

[17/06-17:25:05.728] Application:VoIP

[17/06-17:25:05.729] Sending: [Answering /A21/I456/x0,0/o1], Client ID:0

[17/06-17:25:05.730] Real session ID [456]

[17/06-17:25:05.773] [debug] SetState (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) 2

[17/06-17:25:05.774] [debug] StartConnectionTimer (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) Call from PBX, setting keepalive timer to 90 seconds

[17/06-17:25:05.775] [debug] StartConnectionTimer (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) Started connection timer

[17/06-17:25:05.776] [debug] onConnected UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

[17/06-17:25:05.777] [debug] SetState Set session state to:eConnected

[17/06-17:25:06.193] * Received Packet: ConnectAck /A21/I456/o0

[17/06-17:25:06.194] updateReplyContext: no effect

[17/06-17:25:06.194] Application:VoIP

[17/06-17:25:06.195] MGWConnThread sending: [ConnectAck /A21/I456/o0/#0]

[17/06-17:25:06.198] [debug] ProcessLine received from hgs: [ConnectAck /A21/I456/o0/#0]

[17/06-17:25:06.199] [debug] ProcessLine invoking ConnectAck with tid: 456 cid: 0 params: /A21/I456/o0/#0

[17/06-17:25:06.393] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:09.833] Sending: [ConnectionPing /AMG/I26/S1], Client ID:8

[17/06-17:25:09.838] MGWConnThread sending: [ConnectionPing /AMG/I26/S1/#8]

[17/06-17:25:09.841] [debug] ProcessLine received from hgs: [ConnectionPing /AMG/I26/S1/#8]

[17/06-17:25:09.843] [debug] ProcessLine invoking ConnectionPing with tid: 26 cid: 8 params: /AMG/I26/S1/#8

[17/06-17:25:09.844] Received from MGW: [ConnectionPong /I26/#8]

[17/06-17:25:10.723] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:11.393] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:13.813] * Received Packet: FaultyChannelsInfo /A24/I1204/H5/h14/r31=12,32=13,33=14,34=15

[17/06-17:25:13.814] updateReplyContext: no effect

[17/06-17:25:13.816] Sending: [FaultyChannels /AMG/I87/r12,13,14,15], Client ID:8

[17/06-17:25:13.817] MGWConnThread sending: [FaultyChannels /AMG/I87/r12,13,14,15/#8]

[17/06-17:25:13.823] [debug] ProcessLine received from hgs: [FaultyChannels /AMG/I87/r12,13,14,15/#8]

[17/06-17:25:13.824] [debug] ProcessLine invoking FaultyChannels with tid: 87 cid: 8 params: /AMG/I87/r12,13,14,15/#8

[17/06-17:25:15.723] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:16.393] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:17.614] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 2 NumDigits: 1 HW: 1 TS: 0

[17/06-17:25:17.615] [debug] OnDTMF OnDTMF notification: Digit=2, nHW=14, nTS.Type=0.1

[17/06-17:25:17.963] [debug] exProceedEvent AudioCodes event: EV_BROKEN_RTP_CONNECTION

[17/06-17:25:18.333] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 0 NumDigits: 1 HW: 1 TS: 0

[17/06-17:25:18.335] [debug] OnDTMF OnDTMF notification: Digit=0, nHW=14, nTS.Type=0.1

[17/06-17:25:19.054] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 1 NumDigits: 1 HW: 1 TS: 0

[17/06-17:25:19.055] [debug] OnDTMF OnDTMF notification: Digit=1, nHW=14, nTS.Type=0.1

[17/06-17:25:19.773] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 2 NumDigits: 1 HW: 1 TS: 0

[17/06-17:25:19.775] [debug] OnDTMF OnDTMF notification: Digit=2, nHW=14, nTS.Type=0.1

[17/06-17:25:20.484] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 1 NumDigits: 1 HW: 1 TS: 0

[17/06-17:25:20.485] [debug] OnDTMF OnDTMF notification: Digit=1, nHW=14, nTS.Type=0.1

[17/06-17:25:20.724] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:21.264] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 8 NumDigits: 1 HW: 1 TS: 0

[17/06-17:25:21.265] [debug] OnDTMF OnDTMF notification: Digit=8, nHW=14, nTS.Type=0.1

[17/06-17:25:21.393] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:21.934] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 1 NumDigits: 1 HW: 1 TS: 0

[17/06-17:25:21.935] [debug] OnDTMF OnDTMF notification: Digit=1, nHW=14, nTS.Type=0.1

[17/06-17:25:22.634] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 4 NumDigits: 1 HW: 1 TS: 0

[17/06-17:25:22.635] [debug] OnDTMF OnDTMF notification: Digit=4, nHW=14, nTS.Type=0.1

[17/06-17:25:23.324] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 1 NumDigits: 1 HW: 1 TS: 0

[17/06-17:25:23.327] [debug] OnDTMF OnDTMF notification: Digit=1, nHW=14, nTS.Type=0.1

[17/06-17:25:24.043] * Received Packet: GenericReply /#90/@2b/x0,1/I3033/G

[17/06-17:25:24.074] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 1 NumDigits: 1 HW: 1 TS: 0

[17/06-17:25:24.075] [debug] OnDTMF OnDTMF notification: Digit=1, nHW=14, nTS.Type=0.1

[17/06-17:25:24.204] * Received Packet: SystemTimeEvent /A2b/x0,1/I3034/g11,6,16,5,17,15,56

[17/06-17:25:24.205] Ignoring system time event

[17/06-17:25:24.233] [debug] exProceedEvent AudioCodes event: EV_CONNECTION_ESTABLISHED

[17/06-17:25:25.714] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:26.394] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:28.364] [debug] exProceedEvent AudioCodes event: EV_BROKEN_RTP_CONNECTION

[17/06-17:25:30.004] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 1 NumDigits: 1 HW: 1 TS: 0

[17/06-17:25:30.005] [debug] OnDTMF OnDTMF notification: Digit=1, nHW=14, nTS.Type=0.1

[17/06-17:25:30.724] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:30.725] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 2 NumDigits: 1 HW: 1 TS: 0

[17/06-17:25:30.725] [debug] OnDTMF OnDTMF notification: Digit=2, nHW=14, nTS.Type=0.1

[17/06-17:25:31.394] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:31.464] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 3 NumDigits: 1 HW: 1 TS: 0

[17/06-17:25:31.465] [debug] OnDTMF OnDTMF notification: Digit=3, nHW=14, nTS.Type=0.1

[17/06-17:25:31.695] HMCServer received [ping] from 127.0.0.1:40150

[17/06-17:25:31.695] processRequestLines: reply: "pong " for client: 13

[17/06-17:25:32.174] [debug] exProceedEvent EV_DIGIT: handle: 0 Digit: 4 NumDigits: 1 HW: 1 TS: 0

[17/06-17:25:32.175] [debug] OnDTMF OnDTMF notification: Digit=4, nHW=14, nTS.Type=0.1

[17/06-17:25:32.534] [debug] exProceedEvent AudioCodes event: EV_CONNECTION_ESTABLISHED

[17/06-17:25:35.338] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:25:35.339] [debug] onNewSession UAS:7e61c163@pbxremoteIP: 192.168.1.13

[17/06-17:25:35.340] [debug] SetState Set session state to:eNotConnected

[17/06-17:25:35.340] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:25:35.340] [debug] updateRemoteMediaAddr Updated remote media address to: 192.168.1.13

[17/06-17:25:35.341] [debug] onNewSession UAS:7e61c163@pbxm_TokenToSession.insert

[17/06-17:25:35.342] [debug] CreateIncomingSession token: UAS:7e61c163@pbx remoteNumber: 0012012181444 localNumber: 13109644430 remoteIP: 192.168.1.13

[17/06-17:25:35.342] [debug] Init (/) start session active timer

[17/06-17:25:35.343] [debug] IsCallAllowed Number is not in BlockedDDIs list

[17/06-17:25:35.344] [debug] IsCallAllowed source DDI:13109644430 allowed DDI:^*

[17/06-17:25:35.345] [debug] FindLocalMediaResources (UAS:7e61c163@pbx/) LocalHwyTS: 14:1 RemoteHwyTS: 5:2 media resource: 14.1

[17/06-17:25:35.345] [debug] RegisterSession 1:1 -> UAS:7e61c163@pbx/

[17/06-17:25:35.346] Received from MGW: [Dialing /A21/x0,0/I2/o0/N"0012012181444"/H14/S1/h5/s2/n13109644430/#9]

[17/06-17:25:35.347] packStr=/A21/x0,0/I2/o0/N"0012012181444"/H14/S1/h5/s2/n13109644430

[17/06-17:25:35.347] strHW=14

[17/06-17:25:35.348] strTS=1

[17/06-17:25:35.348] no in filter for add9

[17/06-17:25:35.348] getApplication for:MG.2-VoIP

[17/06-17:25:35.349] strCard=[MG], nClientID=9

[17/06-17:25:35.349] From VoIP Trigger:Dialing /A21/x0,0/I2/o0/N"0012012181444"/H14/S1/h5/s2/n13109644430

[17/06-17:25:35.350] Sending: [Dialing /A21/x0,0/I2/o0/N"0012012181444"/H14/S1/h5/s2/n13109644430], Client ID:9

[17/06-17:25:35.350] No filters required for out address 21.2

[17/06-17:25:35.351] After dial filter: [Dialing /A21/x0,0/I2/o0/N"0012012181444"/H14/S1/h5/s2/n13109644430]

[17/06-17:25:35.351] Real session ID [2]

[17/06-17:25:35.393] [debug] SetState (UAS:7e61c163@pbx/2) 1

[17/06-17:25:35.394] [notice] call from 192.168.1.13 to 13109644430 dialing

[17/06-17:25:35.395] [debug] GetLocalInfo (UAS:7e61c163@pbx/2) GetLocalInfo received RTPAddr:192.168.0.3 wRTPPort:4010

[17/06-17:25:35.396] [debug] OnCreateExternalRTPHandler created external rtp handler board: 192.168.0.3 rtp port: 4010 for token: UAS:7e61c163@pbx

[17/06-17:25:35.397] [debug] onNewSession UAS:UAS:7e61c163@pbx Early Media - Send 183

[17/06-17:25:35.398] [debug] onOffer UAS:7e61c163@pbx

[17/06-17:25:35.399] [debug] provideOkWithSDP UAS: UAS:7e61c163@pbx

[17/06-17:25:35.400] [debug] provideOkWithSDP UAS:offered media:audio|RTP/AVP|0

[17/06-17:25:35.401] [debug] provideOkWithSDP UAS:findFirstMatchingCodecs for: Name=pcmu, payload=0

[17/06-17:25:35.402] [debug] provideOkWithSDP UAS:get strRemoteMediaAddr from msg contents, addr=192.168.1.13

[17/06-17:25:35.402] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:25:35.404] [debug] updateRemoteMediaAddr Updated remote media address to: 192.168.1.13

[17/06-17:25:35.405] [debug] createSdpContents FindMediaReportIP(192.168.1.13)=192.168.1.12

[17/06-17:25:35.406] [debug] createSdpContents addCodec(pcmu,8000)

[17/06-17:25:35.409] [debug] createSdpContents addCodec(telephone-event,8000)

[17/06-17:25:35.410] [debug] provideOkWithSDP provideAnswer

[17/06-17:25:35.411] [debug] onOffer UAS:onOffer - Early media state - send 183. token=UAS:7e61c163@pbx

[17/06-17:25:35.411] [debug] onReadyToSend UAS:7e61c163@pbx

[17/06-17:25:35.412] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:25:35.414] [debug] OnStartExternalRTPHandler start external rtp handler for token: UAS:7e61c163@pbx

[17/06-17:25:35.416] [debug] mediaAllocResources Creating channel with payload 0

[17/06-17:25:35.417] [debug] setChannelparam setting channel parameters board: 0 bus: 1 timeslot: 1 payload: 0 mute DTMF: 0

[17/06-17:25:35.418] [debug] setChannelparam using SIP/RFC2833, payload: 101

[17/06-17:25:35.419] [debug] exOpenChannel Setting mapping: Channel=1 -> HW:TS=1.1

[17/06-17:25:35.432] [debug] exOpenChannel Created channel 1

[17/06-17:25:35.433] [debug] mediaAllocResources OK mediaAllocResources Bus#1 Bus#1 PayLoad=0

[17/06-17:25:35.438] [debug] ~stopwatch mediaAllocResources: 719 usec

[17/06-17:25:35.439] [debug] ~stopwatch openChannel: 772 usec

[17/06-17:25:35.449] [debug] CreateChannel (UAS:7e61c163@pbx/2) created channel timeslot: 1 handle: 1

[17/06-17:25:35.450] [debug] StartMediaStream (UAS:7e61c163@pbx/2) localMediaAddr:192.168.0.3 localMediaPort:4010 remoteMediaAddr:192.168.1.13 remoteMediaPort:48640 Payload:0

[17/06-17:25:35.451] [debug] mediaActivateRTP_RTCPChannel acActivateRTP_RTCPChannel( IPPrec=0, nTOS=0, tx=0,rx=0,ChannelHandle 1 ) returned 0

[17/06-17:25:35.452] [debug] ~stopwatch mediaActivateRTP_RTCPChannel: 288 usec

[17/06-17:25:35.453] [debug] onOffer UAS:Started media, accepting call [uAS:7e61c163@pbx]

[17/06-17:25:35.454] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:35.484] [debug] exProceedEvent AudioCodes event: EV_BROKEN_RTP_CONNECTION

[17/06-17:25:35.514] * Received Packet: DialAck /A21/I2/o1/i1

[17/06-17:25:35.514] updateReplyContext: no effect

[17/06-17:25:35.515] ClientIdMGWSend strMsg=DialAck /A21/I2/o1/i1

[17/06-17:25:35.515] Reply for MGW:DialAck /A21/I2/o1/i1

[17/06-17:25:35.516] MGWConnThread sending: [DialAck /A21/I2/o1/i1/#9]

[17/06-17:25:35.518] [debug] ProcessLine received from hgs: [DialAck /A21/I2/o1/i1/#9]

[17/06-17:25:35.554] [debug] ProcessLine invoking DialAck with tid: 2 cid: 9 params: /A21/I2/o1/i1/#9

[17/06-17:25:35.684] [debug] exProceedEvent AudioCodes event: EV_BROKEN_RTP_CONNECTION

[17/06-17:25:35.724] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:36.394] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:36.395] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:40.434] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:40.724] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:40.725] [debug] exProceedEvent AudioCodes event: EV_CONNECTION_ESTABLISHED

[17/06-17:25:41.394] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:41.395] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:42.414] * Received Packet: Answering /A21/I2/o1

[17/06-17:25:42.415] updateReplyContext: no effect

[17/06-17:25:42.416] ClientIdMGWSend strMsg=Answering /A21/I2/o1

[17/06-17:25:42.417] Reply for MGW:Answering /A21/I2/o1

[17/06-17:25:42.417] MGWConnThread sending: [Answering /A21/I2/o1/#9]

[17/06-17:25:42.422] [debug] ProcessLine received from hgs: [Answering /A21/I2/o1/#9]

[17/06-17:25:42.423] [debug] ProcessLine invoking Answering with tid: 2 cid: 9 params: /A21/I2/o1/#9

[17/06-17:25:42.425] Received from MGW: [ConnectAck /A21/I2/x0,0/o0/#9]

[17/06-17:25:42.427] ClientIdMGWSend strMsg=ConnectAck /A21/I2/x0,0/o0

[17/06-17:25:42.428] Sending: [ConnectAck /A21/I2/x0,0/o0], Client ID:9

[17/06-17:25:42.429] Real session ID [2]

[17/06-17:25:42.473] [debug] SetState (UAS:7e61c163@pbx/2) 2

[17/06-17:25:42.474] [debug] StartConnectionTimer (UAS:7e61c163@pbx/2) Call from Net, setting keepalive timer to 60 seconds

[17/06-17:25:42.475] [debug] StartConnectionTimer (UAS:7e61c163@pbx/2) Started connection timer

[17/06-17:25:42.476] [debug] sendAnswering UAS sendAnswering (UAS:7e61c163@pbx)

[17/06-17:25:42.477] [debug] sendAnswering Sending accept

[17/06-17:25:42.478] [debug] onReadyToSend UAS:7e61c163@pbx

[17/06-17:25:42.479] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:25:42.480] [debug] onConnected UAS:7e61c163@pbx

[17/06-17:25:42.481] [debug] SetState Set session state to:eConnected

[17/06-17:25:42.494] [debug] exProceedEvent AudioCodes event: EV_CONNECTION_ESTABLISHED

[17/06-17:25:42.925] [debug] onReadyToSend UAS:7e61c163@pbx

[17/06-17:25:42.926] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:25:43.914] * Received Packet: GenericReply /#90/@2b/x0,1/I3035/G

[17/06-17:25:43.935] [debug] onReadyToSend UAS:7e61c163@pbx

[17/06-17:25:43.936] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:25:45.434] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:45.724] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:45.945] [debug] onReadyToSend UAS:7e61c163@pbx

[17/06-17:25:45.946] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:25:46.395] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:46.396] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:49.955] [debug] onReadyToSend UAS:7e61c163@pbx

[17/06-17:25:49.956] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:25:50.434] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:50.715] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:51.394] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:51.396] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:52.424] [debug] exProceedEvent AudioCodes event: EV_BROKEN_RTP_CONNECTION

[17/06-17:25:53.635] [debug] exProceedEvent EV_DIGIT: handle: 1 Digit: 3 NumDigits: 1 HW: 1 TS: 1

[17/06-17:25:53.636] [debug] OnDTMF OnDTMF notification: Digit=3, nHW=14, nTS.Type=1.1

[17/06-17:25:53.944] [debug] exProceedEvent EV_DIGIT: handle: 1 Digit: 4 NumDigits: 1 HW: 1 TS: 1

[17/06-17:25:53.946] [debug] OnDTMF OnDTMF notification: Digit=4, nHW=14, nTS.Type=1.1

[17/06-17:25:53.965] [debug] onReadyToSend UAS:7e61c163@pbx

[17/06-17:25:53.966] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:25:55.434] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:55.724] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:56.395] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:56.396] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:25:57.975] [debug] onReadyToSend UAS:7e61c163@pbx

[17/06-17:25:57.976] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:25:59.854] Sending: [ConnectionPing /AMG/I27/S1], Client ID:8

[17/06-17:25:59.855] MGWConnThread sending: [ConnectionPing /AMG/I27/S1/#8]

[17/06-17:25:59.858] [debug] ProcessLine received from hgs: [ConnectionPing /AMG/I27/S1/#8]

[17/06-17:25:59.858] [debug] ProcessLine invoking ConnectionPing with tid: 27 cid: 8 params: /AMG/I27/S1/#8

[17/06-17:25:59.859] Received from MGW: [ConnectionPong /I27/#8]

[17/06-17:26:00.435] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:26:00.435] [debug] exProceedEvent AudioCodes event: EV_CONNECTION_ESTABLISHED

[17/06-17:26:00.725] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:26:01.395] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:26:01.396] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:26:01.685] HMCServer received [ping] from 127.0.0.1:40150

[17/06-17:26:01.686] processRequestLines: reply: "pong " for client: 13

[17/06-17:26:01.985] [debug] onReadyToSend UAS:7e61c163@pbx

[17/06-17:26:01.986] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:26:03.734] * Received Packet: GenericReply /#90/@2b/x0,1/I3036/G

[17/06-17:26:05.094] * Received Packet: FaultyChannelsInfo /A25/I1414/H5/h14/r41=16,42=17,43=18,44=19

[17/06-17:26:05.095] updateReplyContext: no effect

[17/06-17:26:05.096] Sending: [FaultyChannels /AMG/I87/r16,17,18,19], Client ID:8

[17/06-17:26:05.098] MGWConnThread sending: [FaultyChannels /AMG/I87/r16,17,18,19/#8]

[17/06-17:26:05.102] [debug] ProcessLine received from hgs: [FaultyChannels /AMG/I87/r16,17,18,19/#8]

[17/06-17:26:05.103] [debug] ProcessLine invoking FaultyChannels with tid: 87 cid: 8 params: /AMG/I87/r16,17,18,19/#8

[17/06-17:26:05.434] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:26:05.545] [debug] OnSessionActiveTimeout (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) state: 2

[17/06-17:26:05.546] [info] (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) timeslot: 0 handle: 0 call duration: 59 seconds

[17/06-17:26:05.725] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:26:05.825] * Received Packet: FaultyChannelsInfo /A23/I909/H5/h14/r21=8,22=9,23=10,24=11

[17/06-17:26:05.827] updateReplyContext: no effect

[17/06-17:26:05.829] Sending: [FaultyChannels /AMG/I87/r8,9,10,11], Client ID:8

[17/06-17:26:05.830] MGWConnThread sending: [FaultyChannels /AMG/I87/r8,9,10,11/#8]

[17/06-17:26:05.833] [debug] ProcessLine received from hgs: [FaultyChannels /AMG/I87/r8,9,10,11/#8]

[17/06-17:26:05.834] [debug] ProcessLine invoking FaultyChannels with tid: 87 cid: 8 params: /AMG/I87/r8,9,10,11/#8

[17/06-17:26:05.995] [debug] onReadyToSend UAS:7e61c163@pbx

[17/06-17:26:05.996] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:26:06.395] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:26:06.396] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:26:07.774] * Received Packet: KeepAlive /A21/I456/o0

[17/06-17:26:07.775] updateReplyContext: no effect

[17/06-17:26:07.776] Application:VoIP

[17/06-17:26:07.777] MGWConnThread sending: [KeepAlive /A21/I456/o0/#0]

[17/06-17:26:07.780] [debug] ProcessLine received from hgs: [KeepAlive /A21/I456/o0/#0]

[17/06-17:26:07.782] [debug] ProcessLine invoking KeepAlive with tid: 456 cid: 0 params: /A21/I456/o0/#0

[17/06-17:26:07.783] Received from MGW: [KeepAlive /A21/I456/x0,0/o1/#0]

[17/06-17:26:07.784] Application:VoIP

[17/06-17:26:07.785] Sending: [KeepAlive /A21/I456/x0,0/o1], Client ID:0

[17/06-17:26:07.785] Real session ID [456]

[17/06-17:26:10.005] [debug] onReadyToSend UAS:7e61c163@pbx

[17/06-17:26:10.006] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:26:10.435] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:26:10.725] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:26:11.395] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:26:11.395] [debug] exProceedEvent AudioCodes event: EV_MESSAGE_LOG

[17/06-17:26:13.675] * Received Packet: HangingUp /A21/I456/o0/R10

[17/06-17:26:13.675] updateReplyContext: no effect

[17/06-17:26:13.676] Application:VoIP

[17/06-17:26:13.677] ID2App removing ID:456

[17/06-17:26:13.678] MGWConnThread sending: [HangingUp /A21/I456/o0/R10/#0]

[17/06-17:26:13.682] [debug] ProcessLine received from hgs: [HangingUp /A21/I456/o0/R10/#0]

[17/06-17:26:13.683] [debug] ProcessLine invoking HangingUp with tid: 456 cid: 0 params: /A21/I456/o0/R10/#0

[17/06-17:26:13.683] [debug] Hangup (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456)

[17/06-17:26:13.684] [notice] call from 0012012181444 to 8768776075 hangup

[17/06-17:26:13.685] [debug] CloseAudioChannel (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) deactivate rtp channel: 0

[17/06-17:26:13.685] [debug] ~stopwatch mediaDeactivateRTP_RTCPChannel: 177 usec

[17/06-17:26:13.686] [debug] CloseAudioChannel (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) closing channel timeslot: 0 handle: 0

[17/06-17:26:13.686] [debug] mediaCloseResources Closing channel 0

[17/06-17:26:13.687] [debug] ~stopwatch mediaCloseResources: 173 usec

[17/06-17:26:13.687] [debug] ~stopwatch closeChannel: 282 usec

[17/06-17:26:13.688] [debug] Hangup (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) waiting for call statistics...

[17/06-17:26:13.697] [debug] exProceedEvent received acEV_RTCP_CLOSE

[17/06-17:26:13.698] [debug] OnCloseRTP (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456)

[17/06-17:26:13.699] [debug] UnregisterSession 1:0 -> UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456

[17/06-17:26:13.699] [debug] Close (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456)

[17/06-17:26:13.700] [debug] Close (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) closing session timeslot:0 handle: -1

[17/06-17:26:13.701] [debug] CloseResources (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456)

[17/06-17:26:13.707] [debug] CloseAudioChannel (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) channel timeslot: 0 already closed

[17/06-17:26:13.708] [debug] CloseResources (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) remove media resoruce: 14.0

[17/06-17:26:13.709] [debug] DisconnectEndpoints (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456)

[17/06-17:26:13.710] [debug] ClearNetCall (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456) Cause Code: 10 Converted Cause Code: 10

[17/06-17:26:13.711] [debug] UpdateStatsAndCDR (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456)

[17/06-17:26:13.712] [info] VoIP CDR: 4,2011-06-17 T 17:25:05,2011-06-17 T 17:25:05,0012012181444,8768776075,2011-06-17 T 17:25:05,2011-06-17 T 17:26:13,67,192.168.1.13,192.168.1.13,53830,1,0,1,1,3403,935,0,10

[17/06-17:26:13.713] [debug] ~tMGWSession (UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY./456)

[17/06-17:26:13.715] [debug] clearCall ClearCall [uAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.]

[17/06-17:26:13.716] [debug] setHangupReason m_nHangupQ931: 10 m_nHangupSIP: 480

[17/06-17:26:13.716] [debug] endCall Q931 reason: 10 SIP reason: 480

[17/06-17:26:13.717] [debug] onReadyToSend UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

[17/06-17:26:13.718] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:26:13.719] [debug] onTerminated UAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.] reason[1]

[17/06-17:26:13.737] [debug] callTeardown telephony Disconnected

[17/06-17:26:13.740] [debug] removeSession Call Id [uAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.]: m_tokenToSession.erase

[17/06-17:26:13.741] [debug] ~tSessionInfo Call id [uAC:YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.]: Calling ~tSessionInfo

[17/06-17:26:14.016] [debug] onReadyToSend UAS:7e61c163@pbx

[17/06-17:26:14.017] [debug] getRemoteSignalIP getRemoteSignalIP=192.168.1.13

[17/06-17:26:14.426] [debug] onAckNotReceived UAS:

[17/06-17:26:14.428] [debug] callTeardown network Disconnected

[17/06-17:26:14.429] [debug] OnRemoteNetDisconnect reason: 10 for token: UAS:7e61c163@pbx

[17/06-17:26:14.429] [debug] Hangup (UAS:7e61c163@pbx/2)

[17/06-17:26:14.430] [notice] call from 0012012181444 to 13109644430 hangup

[17/06-17:26:14.431] [debug] CloseAudioChannel (UAS:7e61c163@pbx/2) deactivate rtp channel: 1

[17/06-17:26:14.432] [debug] ~stopwatch mediaDeactivateRTP_RTCPChannel: 181 usec

[17/06-17:26:14.433] [debug] CloseAudioChannel (UAS:7e61c163@pbx/2) closing channel timeslot: 1 handle: 1

[17/06-17:26:14.433] [debug] mediaCloseResources Closing channel 1

[17/06-17:26:14.437] [debug] ~stopwatch mediaCloseResources: 185 usec

[17/06-17:26:14.438] [debug] ~stopwatch closeChannel: 293 usec

[17/06-17:26:14.439] [debug] Hangup (UAS:7e61c163@pbx/2) waiting for call statistics...

[17/06-17:26:14.440] [debug] removeSession Call Id [uAS:7e61c163@pbx]: m_tokenToSession.erase

[17/06-17:26:14.441] [debug] ~tSessionInfo Call id [uAS:7e61c163@pbx]: Calling ~tSessionInfo

[17/06-17:26:14.443] [debug] exProceedEvent received acEV_RTCP_CLOSE

[17/06-17:26:14.444] [debug] OnCloseRTP (UAS:7e61c163@pbx/2)

[17/06-17:26:14.445] [debug] UnregisterSession 1:1 -> UAS:7e61c163@pbx/2

[17/06-17:26:14.446] [debug] Close (UAS:7e61c163@pbx/2)

[17/06-17:26:14.446] [debug] Close (UAS:7e61c163@pbx/2) closing session timeslot:1 handle: -1

[17/06-17:26:14.447] [debug] CloseResources (UAS:7e61c163@pbx/2)

[17/06-17:26:14.448] [debug] CloseAudioChannel (UAS:7e61c163@pbx/2) channel timeslot: 1 already closed

[17/06-17:26:14.449] [debug] CloseResources (UAS:7e61c163@pbx/2) remove media resoruce: 14.1

[17/06-17:26:14.450] [debug] DisconnectEndpoints (UAS:7e61c163@pbx/2)

[17/06-17:26:14.451] [debug] SendHangupToCard Cause Code =10 (SendHangupToCard)

[17/06-17:26:14.452] [debug] SendHangupToCard Converted Cause Code=10

[17/06-17:26:14.453] Received from MGW: [HangingUp /A21/I2/x0,0/o0/R10/#9]

[17/06-17:26:14.455] ClientIdMGWSend strMsg=HangingUp /A21/I2/x0,0/o0/R10

[17/06-17:26:14.456] Sending: [HangingUp /A21/I2/x0,0/o0/R10], Client ID:9

[17/06-17:26:14.457] Real session ID [2]

[17/06-17:26:14.504] [debug] UpdateStatsAndCDR (UAS:7e61c163@pbx/2)

[17/06-17:26:14.505] [info] VoIP CDR: 5,2011-06-17 T 17:25:35,2011-06-17 T 17:25:35,0012012181444,13109644430,2011-06-17 T 17:25:42,2011-06-17 T 17:26:14,32,192.168.1.13,192.168.1.13,48640,2,0,0,0,1954,484,0,10

[17/06-17:26:14.506] [debug] ~tMGWSession (UAS:7e61c163@pbx/2)

[17/06-17:26:14.775] * Received Packet: ClearAck /A21/I2/o1

[17/06-17:26:14.775] updateReplyContext: no effect

[17/06-17:26:14.776] ClientIdMGWSend strMsg=ClearAck /A21/I2/o1

[17/06-17:26:14.776] ID2App removing ID:2

[17/06-17:26:14.777] Reply for MGW:ClearAck /A21/I2/o1

[17/06-17:26:14.777] MGWConnThread sending: [ClearAck /A21/I2/o1/#9]

[17/06-17:26:14.780] [debug] ProcessLine received from hgs: [ClearAck /A21/I2/o1/#9]

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Here is the corresponding PBX Log when Hg4000 transport = TCP. I hope this can be resolved either by getting PBX to work when Hypermedia transport=UDP or force PBX transport = TCP...

 

INVITE sip:8768776075@192.168.1.13 SIP/2.0

Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-dc12fe7d8c7cfa32-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.12:5060>

To: <sip:8768776075@192.168.1.13>

From: <sip:0012012181444@192.168.1.12:5060>;tag=ab429725

Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub, 100rel, em

User-Agent: HG4000/1.0

Content-Length: 345

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4000 RTP/AVP 4 18 18 18 18 0 8 101

a=rtpmap:4 G723/8000

a=rtpmap:18 G729/8000

a=rtpmap:18 G729a/8000

a=rtpmap:18 G729b/8000

a=rtpmap:18 G729ab/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110616172433: SIP Tx tcp:192.168.1.12:12330:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-dc12fe7d8c7cfa32-1---d8754z-;rport=12330

From: <sip:0012012181444@192.168.1.12:5060>;tag=ab429725

To: <sip:8768776075@192.168.1.13>;tag=bbfe18e7ec

Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

CSeq: 1 INVITE

Content-Length: 0

 

 

[5] 20110616172433: Using <sip:0012012181444@192.168.1.12:5060;user=phone> as redirect source address

[5] 20110616172433: SIP Tx tcp:192.168.1.13:6060:

INVITE sip:8768776075@192.168.1.13:6060;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:65150;branch=z9hG4bK-4a388cfccf9a8ae060fbe113e148ad53;rport

From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

To: <sip:8768776075@192.168.1.13:6060;user=phone>

Call-ID: 99529a35@pbx

CSeq: 1767 INVITE

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:65150;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

Related-Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Type: application/sdp

Content-Length: 327

 

v=0

o=- 17834 17834 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 36412 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110616172433: SIP Rx tcp:192.168.1.13:6060:

SIP/2.0 100 Trying

FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

TO: <sip:8768776075@192.168.1.13:6060;user=phone>

CSEQ: 1767 INVITE

CALL-ID: 99529a35@pbx

VIA: SIP/2.0/TCP 192.168.1.13:65150;branch=z9hG4bK-4a388cfccf9a8ae060fbe113e148ad53;rport

CONTENT-LENGTH: 0

 

 

[5] 20110616172433: SIP Tx tcp:192.168.1.12:12330:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-dc12fe7d8c7cfa32-1---d8754z-;rport=12330

From: <sip:0012012181444@192.168.1.12:5060>;tag=ab429725

To: <sip:8768776075@192.168.1.13>;tag=bbfe18e7ec

Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

CSeq: 1 INVITE

Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 253

 

v=0

o=- 28006 28006 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 54788 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110616172433: SIP Rx tcp:192.168.1.13:6060:

SIP/2.0 302 Moved Temporarily

FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

TO: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=1652901e79

CSEQ: 1767 INVITE

CALL-ID: 99529a35@pbx

VIA: SIP/2.0/TCP 192.168.1.13:65150;branch=z9hG4bK-4a388cfccf9a8ae060fbe113e148ad53;rport

CONTACT: <sip:8768776075@192.168.1.13:65122;user=phone;transport=Tcp;maddr=192.168.1.13;x-mss-call-id=99529a35%40pbx>

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[7] 20110616172433: Call 99529a35@pbx: Clear last INVITE

[5] 20110616172433: SIP Tx tcp:192.168.1.13:6060:

ACK sip:8768776075@192.168.1.13:6060;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:65150;branch=z9hG4bK-4a388cfccf9a8ae060fbe113e148ad53;rport

From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=1652901e79

Call-ID: 99529a35@pbx

CSeq: 1767 ACK

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:65150;transport=tcp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110616172433: Redirecting call

[5] 20110616172433: SIP Tx tcp:192.168.1.13:65122:

INVITE sip:8768776075@192.168.1.13:65122;user=phone;transport=Tcp;maddr=192.168.1.13;x-mss-call-id=99529a35%40pbx SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-9c613fdf39873449343bbb52519440be;rport

From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

To: <sip:8768776075@192.168.1.13:6060;user=phone>

Call-ID: 99529a35@pbx

CSeq: 1768 INVITE

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:65152;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

Related-Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Type: application/sdp

Content-Length: 327

 

v=0

o=- 17834 17834 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 36412 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110616172433: SIP Rx tcp:192.168.1.13:65122:

SIP/2.0 100 Trying

FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

TO: <sip:8768776075@192.168.1.13:6060;user=phone>

CSEQ: 1768 INVITE

CALL-ID: 99529a35@pbx

VIA: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-9c613fdf39873449343bbb52519440be;rport

CONTENT-LENGTH: 0

 

 

[5] 20110616172433: SIP Rx tcp:192.168.1.13:65122:

SIP/2.0 180 Ringing

FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

TO: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=834EFDD15A;tag=49062df6

CSEQ: 1768 INVITE

CALL-ID: 99529a35@pbx

VIA: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-9c613fdf39873449343bbb52519440be;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[5] 20110616172433: SIP Rx tcp:192.168.1.13:65122:

SIP/2.0 200 OK

FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

TO: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=834EFDD15A;tag=49062df6

CSEQ: 1768 INVITE

CALL-ID: 99529a35@pbx

VIA: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-9c613fdf39873449343bbb52519440be;rport

CONTACT: <sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13>;automata

CONTENT-LENGTH: 194

CONTENT-TYPE: application/sdp

ALLOW: UPDATE

SERVER: RTCC/3.0.0.0

ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

 

v=0

o=- 0 0 IN IP4 192.168.1.13

s=Microsoft Speech Server session

c=IN IP4 192.168.1.13

t=0 0

m=audio 38400 RTP/AVP 0 8 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

 

[7] 20110616172433: Call 99529a35@pbx: Clear last INVITE

[5] 20110616172433: SIP Tx tcp:192.168.1.13:65122:

ACK sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13 SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-ba37c71e815d8926de016156b10dbff6;rport

From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=49062df6;epid=834EFDD15A

Call-ID: 99529a35@pbx

CSeq: 1768 ACK

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:65152;transport=tcp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110616172433: SIP Tx tcp:192.168.1.12:12330:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-dc12fe7d8c7cfa32-1---d8754z-;rport=12330

From: <sip:0012012181444@192.168.1.12:5060>;tag=ab429725

To: <sip:8768776075@192.168.1.13>;tag=bbfe18e7ec

Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

CSeq: 1 INVITE

Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Content-Type: application/sdp

Content-Length: 253

 

v=0

o=- 28006 28006 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 54788 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110616172433: SIP Rx tcp:192.168.1.12:12330:

ACK sip:8768776075@192.168.1.13:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-ebdfbe3d0bdb584f-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.12:5060>

To: <sip:8768776075@192.168.1.13>;tag=bbfe18e7ec

From: <sip:0012012181444@192.168.1.12:5060>;tag=ab429725

Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

CSeq: 1 ACK

User-Agent: HG4000/1.0

Content-Length: 0

 

 

[5] 20110616172441: SIP Rx tcp:192.168.1.12:12330:

BYE sip:8768776075@192.168.1.13:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-b83b342fda56d142-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.12:5060>

To: <sip:8768776075@192.168.1.13>;tag=bbfe18e7ec

From: <sip:0012012181444@192.168.1.12:5060>;tag=ab429725

Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

CSeq: 2 BYE

User-Agent: HG4000/1.0

Reason: SIP;description="ACK not received"

Content-Length: 0

 

 

[5] 20110616172441: SIP Tx tcp:192.168.1.12:12330:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-b83b342fda56d142-1---d8754z-;rport=12330;received=192.168.1.12

From: <sip:0012012181444@192.168.1.12:5060>;tag=ab429725

To: <sip:8768776075@192.168.1.13>;tag=bbfe18e7ec

Call-ID: MzU1ZjBkMjcwMzEzZmIyYzQ1ZGZjMDYxZTEzNTYzNGE.

CSeq: 2 BYE

Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

User-Agent: snom-PBX/2011-4.2.1.4009

Content-Length: 0

 

 

[5] 20110616172441: SIP Tx tcp:192.168.1.13:65122:

BYE sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13 SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-897f7badaf3d7ad46c5b67c81342b878;rport

From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=49062df6

Call-ID: 99529a35@pbx

CSeq: 1769 BYE

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:65152;transport=tcp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110616172441: SIP Rx tcp:192.168.1.13:65122:

SIP/2.0 200 OK

FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=31288

TO: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=49062df6;epid=834EFDD15A

CSEQ: 1769 BYE

CALL-ID: 99529a35@pbx

VIA: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-897f7badaf3d7ad46c5b67c81342b878;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[7] 20110616172441: Call 99529a35@pbx: Clear last request

[5] 20110616172441: BYE Response: Terminate 99529a35@pbx

[5] 20110616172442: SIP Rx udp:192.168.1.13:41032:

SUBSCRIBE sip:1000@192.168.1.13 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:41032;branch=z9hG4bK-d8754z-eadee8af60c2b8cf-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:1000@192.168.1.13:41032>

To: "1000"<sip:1000@192.168.1.13>

From: "1000"<sip:1000@192.168.1.13>;tag=ba371dbd

Call-ID: ZjE0YjMyMmVjYWNiOGZkZTkxOGIyZTE3ODhlYWY1YzU.

CSeq: 1 SUBSCRIBE

Expires: 300

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

User-Agent: X-Lite 4 release 4.0 stamp 58832

Event: message-summary

Content-Length: 0

 

 

[5] 20110616172442: SIP Tx udp:192.168.1.13:41032:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.1.13:41032;branch=z9hG4bK-d8754z-eadee8af60c2b8cf-1---d8754z-;rport=41032

From: "1000" <sip:1000@192.168.1.13>;tag=ba371dbd

To: "1000" <sip:1000@192.168.1.13>;tag=ba7af3f391

Call-ID: ZjE0YjMyMmVjYWNiOGZkZTkxOGIyZTE3ODhlYWY1YzU.

CSeq: 1 SUBSCRIBE

Content-Length: 0

 

 

[5] 20110616172502: SIP Rx tcp:192.168.1.12:12330:

INVITE sip:8768776075@192.168.1.13 SIP/2.0

Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-450bb10b9ca6da46-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.12:5060>

To: <sip:8768776075@192.168.1.13>

From: <sip:0012012181444@192.168.1.12:5060>;tag=29823d4a

Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub, 100rel, em

User-Agent: HG4000/1.0

Content-Length: 345

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4000 RTP/AVP 4 18 18 18 18 0 8 101

a=rtpmap:4 G723/8000

a=rtpmap:18 G729/8000

a=rtpmap:18 G729a/8000

a=rtpmap:18 G729b/8000

a=rtpmap:18 G729ab/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110616172502: SIP Tx tcp:192.168.1.12:12330:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-450bb10b9ca6da46-1---d8754z-;rport=12330

From: <sip:0012012181444@192.168.1.12:5060>;tag=29823d4a

To: <sip:8768776075@192.168.1.13>;tag=92de13d950

Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

CSeq: 1 INVITE

Content-Length: 0

 

 

[5] 20110616172502: Using <sip:0012012181444@192.168.1.12:5060;user=phone> as redirect source address

[5] 20110616172502: SIP Tx tcp:192.168.1.13:6060:

INVITE sip:8768776075@192.168.1.13:6060;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:65150;branch=z9hG4bK-ddf022c07713811d4d214d87b8100b76;rport

From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

To: <sip:8768776075@192.168.1.13:6060;user=phone>

Call-ID: 5d084739@pbx

CSeq: 16994 INVITE

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:65150;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

Related-Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Type: application/sdp

Content-Length: 327

 

v=0

o=- 38841 38841 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 49044 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110616172502: SIP Rx tcp:192.168.1.13:6060:

SIP/2.0 100 Trying

FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

TO: <sip:8768776075@192.168.1.13:6060;user=phone>

CSEQ: 16994 INVITE

CALL-ID: 5d084739@pbx

VIA: SIP/2.0/TCP 192.168.1.13:65150;branch=z9hG4bK-ddf022c07713811d4d214d87b8100b76;rport

CONTENT-LENGTH: 0

 

 

[5] 20110616172502: SIP Tx tcp:192.168.1.12:12330:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-450bb10b9ca6da46-1---d8754z-;rport=12330

From: <sip:0012012181444@192.168.1.12:5060>;tag=29823d4a

To: <sip:8768776075@192.168.1.13>;tag=92de13d950

Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

CSeq: 1 INVITE

Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 253

 

v=0

o=- 57347 57347 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 53830 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110616172502: SIP Rx tcp:192.168.1.13:6060:

SIP/2.0 302 Moved Temporarily

FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

TO: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=7a5d8de81d

CSEQ: 16994 INVITE

CALL-ID: 5d084739@pbx

VIA: SIP/2.0/TCP 192.168.1.13:65150;branch=z9hG4bK-ddf022c07713811d4d214d87b8100b76;rport

CONTACT: <sip:8768776075@192.168.1.13:65122;user=phone;transport=Tcp;maddr=192.168.1.13;x-mss-call-id=5d084739%40pbx>

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[7] 20110616172502: Call 5d084739@pbx: Clear last INVITE

[5] 20110616172502: SIP Tx tcp:192.168.1.13:6060:

ACK sip:8768776075@192.168.1.13:6060;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:65150;branch=z9hG4bK-ddf022c07713811d4d214d87b8100b76;rport

From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=7a5d8de81d

Call-ID: 5d084739@pbx

CSeq: 16994 ACK

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:65150;transport=tcp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110616172502: Redirecting call

[5] 20110616172502: SIP Tx tcp:192.168.1.13:65122:

INVITE sip:8768776075@192.168.1.13:65122;user=phone;transport=Tcp;maddr=192.168.1.13;x-mss-call-id=5d084739%40pbx SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-25324b29831544a6365c1bd4b82aa8e5;rport

From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

To: <sip:8768776075@192.168.1.13:6060;user=phone>

Call-ID: 5d084739@pbx

CSeq: 16995 INVITE

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:65152;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

Related-Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Type: application/sdp

Content-Length: 327

 

v=0

o=- 38841 38841 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 49044 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110616172502: SIP Rx tcp:192.168.1.13:65122:

SIP/2.0 100 Trying

FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

TO: <sip:8768776075@192.168.1.13:6060;user=phone>

CSEQ: 16995 INVITE

CALL-ID: 5d084739@pbx

VIA: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-25324b29831544a6365c1bd4b82aa8e5;rport

CONTENT-LENGTH: 0

 

 

[5] 20110616172502: SIP Rx tcp:192.168.1.13:65122:

SIP/2.0 180 Ringing

FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

TO: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=834EFDD15A;tag=bb7e72d89

CSEQ: 16995 INVITE

CALL-ID: 5d084739@pbx

VIA: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-25324b29831544a6365c1bd4b82aa8e5;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[5] 20110616172502: SIP Rx tcp:192.168.1.13:65122:

SIP/2.0 200 OK

FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

TO: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=834EFDD15A;tag=bb7e72d89

CSEQ: 16995 INVITE

CALL-ID: 5d084739@pbx

VIA: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-25324b29831544a6365c1bd4b82aa8e5;rport

CONTACT: <sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13>;automata

CONTENT-LENGTH: 194

CONTENT-TYPE: application/sdp

ALLOW: UPDATE

SERVER: RTCC/3.0.0.0

ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

 

v=0

o=- 0 0 IN IP4 192.168.1.13

s=Microsoft Speech Server session

c=IN IP4 192.168.1.13

t=0 0

m=audio 50944 RTP/AVP 0 8 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

 

[7] 20110616172502: Call 5d084739@pbx: Clear last INVITE

[5] 20110616172502: SIP Tx tcp:192.168.1.13:65122:

ACK sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13 SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-eee5763cf0213961f4bf7dd0f5495d8e;rport

From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=bb7e72d89;epid=834EFDD15A

Call-ID: 5d084739@pbx

CSeq: 16995 ACK

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:65152;transport=tcp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110616172502: SIP Tx tcp:192.168.1.12:12330:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.1.12:5060;branch=z9hG4bK-d8754z-450bb10b9ca6da46-1---d8754z-;rport=12330

From: <sip:0012012181444@192.168.1.12:5060>;tag=29823d4a

To: <sip:8768776075@192.168.1.13>;tag=92de13d950

Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

CSeq: 1 INVITE

Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Content-Type: application/sdp

Content-Length: 253

 

v=0

o=- 57347 57347 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 53830 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110616172502: SIP Rx tcp:192.168.1.12:12330:

ACK sip:8768776075@192.168.1.13:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-90e6572bc2e79711-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.12:5060>

To: <sip:8768776075@192.168.1.13>;tag=92de13d950

From: <sip:0012012181444@192.168.1.12:5060>;tag=29823d4a

Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

CSeq: 1 ACK

User-Agent: HG4000/1.0

Content-Length: 0

 

 

[5] 20110616172531: SIP Rx tcp:192.168.1.13:65122:

INVITE sip:0012012181444@192.168.1.13:65152;transport=tcp SIP/2.0

FROM: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=834EFDD15A;tag=bb7e72d89

TO: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

CSEQ: 1 INVITE

CALL-ID: 5d084739@pbx

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.1.13:65122;branch=z9hG4bK3ffd794d

CONTACT: <sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13;ms-opaque=e6d2caedfe360c75>;automata

CONTENT-LENGTH: 206

USER-AGENT: RTCC/3.0.0.0

CONTENT-TYPE: application/sdp

 

v=0

o=- 0 0 IN IP4 192.168.1.13

s=Microsoft Speech Server session

c=IN IP4 192.168.1.13

t=0 0

m=audio 50944 RTP/AVP 0 8 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendonly

a=ptime:20

 

[5] 20110616172531: SIP Tx tcp:192.168.1.13:65122:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.1.13:65122;branch=z9hG4bK3ffd794d

From: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=bb7e72d89;epid=834EFDD15A

To: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

Call-ID: 5d084739@pbx

CSeq: 1 INVITE

Contact: <sip:0012012181444@192.168.1.13:65152;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Content-Type: application/sdp

Content-Length: 265

 

v=0

o=- 38841 38841 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 49044 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=recvonly

 

[5] 20110616172531: SIP Rx tcp:192.168.1.13:65122:

ACK sip:0012012181444@192.168.1.13:65152;transport=tcp SIP/2.0

FROM: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=834EFDD15A;tag=bb7e72d89

TO: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

CSEQ: 1 ACK

CALL-ID: 5d084739@pbx

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.1.13:65122;branch=z9hG4bKc5459f2d

CONTENT-LENGTH: 0

USER-AGENT: RTCC/3.0.0.0

 

 

[5] 20110616172531: SIP Rx tcp:192.168.1.13:65159:

INVITE sip:13109644430@192.168.1.13:5060;transport=tcp SIP/2.0

FROM: <sip:0012012181444@192.168.1.12:5060;transport=tcp>;epid=834EFDD15A;tag=d01782b7fb

TO: <sip:13109644430@192.168.1.13:5060;transport=tcp>

CSEQ: 2 INVITE

CALL-ID: 9b5e977d-efb2-4bf6-99e7-b07c4e631f0c

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.1.13:65159;branch=z9hG4bKf27d5133

CONTACT: <sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13;ms-opaque=e6d2caedfe360c75>;automata

CONTENT-LENGTH: 336

USER-AGENT: RTCC/3.0.0.0

CONTENT-TYPE: application/sdp

ALLOW: UPDATE

ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

 

v=0

o=- 0 0 IN IP4 192.168.1.13

s=Microsoft Speech Server session

c=IN IP4 192.168.1.13

t=0 0

m=audio 38400 RTP/AVP 114 115 4 0 8 97 101

a=rtpmap:114 x-msrta/16000

a=fmtp:114 bitrate=29000

a=rtpmap:115 x-msrta/8000

a=fmtp:115 bitrate=11800

a=rtpmap:97 RED/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

 

[5] 20110616172531: SIP Tx tcp:192.168.1.13:65159:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 192.168.1.13:65159;branch=z9hG4bKf27d5133

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp>;tag=d01782b7fb;epid=834EFDD15A

To: <sip:13109644430@192.168.1.13:5060;transport=tcp>;tag=51fdee39e8

Call-ID: 9b5e977d-efb2-4bf6-99e7-b07c4e631f0c

CSeq: 2 INVITE

Content-Length: 0

 

 

[5] 20110616172531: Using <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone> as redirect source address

[5] 20110616172531: SIP Tx tcp:192.168.1.12:5060:

INVITE sip:13109644430@192.168.1.12;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

To: <sip:13109644430@192.168.1.12;user=phone>

Call-ID: 7e61c163@pbx

CSeq: 19524 INVITE

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:65160;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Diversion: <tel:42>;reason=unconditional;screen=no;privacy=off

Related-Call-ID: 9b5e977d-efb2-4bf6-99e7-b07c4e631f0c

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Type: application/sdp

Content-Length: 327

 

v=0

o=- 44371 44371 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 48640 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110616172531: SIP Tx tcp:192.168.1.13:65159:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TCP 192.168.1.13:65159;branch=z9hG4bKf27d5133

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp>;tag=d01782b7fb;epid=834EFDD15A

To: <sip:13109644430@192.168.1.13:5060;transport=tcp>;tag=51fdee39e8

Call-ID: 9b5e977d-efb2-4bf6-99e7-b07c4e631f0c

CSeq: 2 INVITE

Contact: <sip:13109644430@192.168.1.13:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Content-Type: application/sdp

Content-Length: 263

 

v=0

o=- 8882 8882 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 32724 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110616172532: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

Call-ID: 7e61c163@pbx

CSeq: 19524 INVITE

Content-Type: application/sdp

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110616172539: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

Call-ID: 7e61c163@pbx

CSeq: 19524 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[7] 20110616172539: Call 7e61c163@pbx: Clear last INVITE

[5] 20110616172539: SIP Tx udp:192.168.1.12:5060:

ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

Call-ID: 7e61c163@pbx

CSeq: 19524 ACK

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110616172539: SIP Tx tcp:192.168.1.13:65159:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.1.13:65159;branch=z9hG4bKf27d5133

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp>;tag=d01782b7fb;epid=834EFDD15A

To: <sip:13109644430@192.168.1.13:5060;transport=tcp>;tag=51fdee39e8

Call-ID: 9b5e977d-efb2-4bf6-99e7-b07c4e631f0c

CSeq: 2 INVITE

Contact: <sip:13109644430@192.168.1.13:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4009

Content-Type: application/sdp

Content-Length: 263

 

v=0

o=- 8882 8882 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 32724 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

 

[5] 20110616172539: SIP Rx tcp:192.168.1.13:65159:

ACK sip:13109644430@192.168.1.13:5060;transport=tcp SIP/2.0

FROM: <sip:0012012181444@192.168.1.12:5060;transport=tcp>;epid=834EFDD15A;tag=d01782b7fb

TO: <sip:13109644430@192.168.1.13:5060;transport=tcp>;tag=51fdee39e8

CSEQ: 2 ACK

CALL-ID: 9b5e977d-efb2-4bf6-99e7-b07c4e631f0c

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.1.13:65159;branch=z9hG4bKb96c31c1

CONTENT-LENGTH: 0

USER-AGENT: RTCC/3.0.0.0

 

 

[5] 20110616172539: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

Call-ID: 7e61c163@pbx

CSeq: 19524 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110616172539: SIP Tx udp:192.168.1.12:5060:

ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

Call-ID: 7e61c163@pbx

CSeq: 19524 ACK

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110616172540: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

Call-ID: 7e61c163@pbx

CSeq: 19524 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110616172540: SIP Tx udp:192.168.1.12:5060:

ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

Call-ID: 7e61c163@pbx

CSeq: 19524 ACK

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110616172542: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

Call-ID: 7e61c163@pbx

CSeq: 19524 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110616172542: SIP Tx udp:192.168.1.12:5060:

ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

Call-ID: 7e61c163@pbx

CSeq: 19524 ACK

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110616172546: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

Call-ID: 7e61c163@pbx

CSeq: 19524 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110616172546: SIP Tx udp:192.168.1.12:5060:

ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

Call-ID: 7e61c163@pbx

CSeq: 19524 ACK

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110616172550: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

Call-ID: 7e61c163@pbx

CSeq: 19524 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110616172550: SIP Tx udp:192.168.1.12:5060:

ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

Call-ID: 7e61c163@pbx

CSeq: 19524 ACK

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110616172554: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

Call-ID: 7e61c163@pbx

CSeq: 19524 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110616172554: SIP Tx udp:192.168.1.12:5060:

ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

Call-ID: 7e61c163@pbx

CSeq: 19524 ACK

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110616172558: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

Call-ID: 7e61c163@pbx

CSeq: 19524 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110616172558: SIP Tx udp:192.168.1.12:5060:

ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

Call-ID: 7e61c163@pbx

CSeq: 19524 ACK

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[6] 20110616172602: SIP TCP/TLS timeout on 192.168.1.13:6060, closing connection

[5] 20110616172602: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

Call-ID: 7e61c163@pbx

CSeq: 19524 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110616172602: SIP Tx udp:192.168.1.12:5060:

ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

Call-ID: 7e61c163@pbx

CSeq: 19524 ACK

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110616172606: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

Call-ID: 7e61c163@pbx

CSeq: 19524 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110616172606: SIP Tx udp:192.168.1.12:5060:

ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

Call-ID: 7e61c163@pbx

CSeq: 19524 ACK

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110616172610: SIP Rx tcp:192.168.1.12:12330:

BYE sip:8768776075@192.168.1.13:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-48372c12e9252c06-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.12:5060>

To: <sip:8768776075@192.168.1.13>;tag=92de13d950

From: <sip:0012012181444@192.168.1.12:5060>;tag=29823d4a

Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

CSeq: 2 BYE

User-Agent: HG4000/1.0

Reason: SIP;description="ACK not received"

Content-Length: 0

 

 

[5] 20110616172610: SIP Tx tcp:192.168.1.12:12330:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.0.2:5060;branch=z9hG4bK-d8754z-48372c12e9252c06-1---d8754z-;rport=12330;received=192.168.1.12

From: <sip:0012012181444@192.168.1.12:5060>;tag=29823d4a

To: <sip:8768776075@192.168.1.13>;tag=92de13d950

Call-ID: YjYxOTUxNWRiMTdhZDExYjlkNTI2MTYyNzMwYzRmYmY.

CSeq: 2 BYE

Contact: <sip:8768776075@192.168.1.13:5060;transport=tcp>

User-Agent: snom-PBX/2011-4.2.1.4009

Content-Length: 0

 

 

[5] 20110616172610: SIP Tx tcp:192.168.1.13:65122:

BYE sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13 SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-cd85a3105db01a4a2724cb991e7945f2;rport

From: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

To: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=bb7e72d89

Call-ID: 5d084739@pbx

CSeq: 16996 BYE

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:65152;transport=tcp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110616172610: SIP Rx tcp:192.168.1.13:65122:

SIP/2.0 200 OK

FROM: <sip:0012012181444@192.168.1.12:5060;user=phone>;tag=4132

TO: <sip:8768776075@192.168.1.13:6060;user=phone>;tag=bb7e72d89;epid=834EFDD15A

CSEQ: 16996 BYE

CALL-ID: 5d084739@pbx

VIA: SIP/2.0/TCP 192.168.1.13:65152;branch=z9hG4bK-cd85a3105db01a4a2724cb991e7945f2;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[7] 20110616172610: Call 5d084739@pbx: Clear last request

[5] 20110616172610: BYE Response: Terminate 5d084739@pbx

[5] 20110616172610: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

Call-ID: 7e61c163@pbx

CSeq: 19524 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110616172610: SIP Tx udp:192.168.1.12:5060:

ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

Call-ID: 7e61c163@pbx

CSeq: 19524 ACK

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[6] 20110616172710: SIP TCP/TLS timeout on 192.168.1.13:65122, closing connection

[6] 20110616172710: SIP TCP/TLS timeout on 192.168.1.12:5060, closing connection

[5] 20110616172745: SIP Rx udp:192.168.1.13:41032:

SUBSCRIBE sip:1000@192.168.1.13 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:41032;branch=z9hG4bK-d8754z-f4f6dcd533171c22-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:1000@192.168.1.13:41032>

To: "1000"<sip:1000@192.168.1.13>

From: "1000"<sip:1000@192.168.1.13>;tag=565b18cf

Call-ID: MDY2NDEzMDgzOTlmMTM1OWI0ZjNiOTFjNGE4MDZmMTA.

CSeq: 1 SUBSCRIBE

Expires: 300

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

User-Agent: X-Lite 4 release 4.0 stamp 58832

Event: message-summary

Content-Length: 0

 

 

[5] 20110616172745: SIP Tx udp:192.168.1.13:41032:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.1.13:41032;branch=z9hG4bK-d8754z-f4f6dcd533171c22-1---d8754z-;rport=41032

From: "1000" <sip:1000@192.168.1.13>;tag=565b18cf

To: "1000" <sip:1000@192.168.1.13>;tag=badc876e36

Call-ID: MDY2NDEzMDgzOTlmMTM1OWI0ZjNiOTFjNGE4MDZmMTA.

CSeq: 1 SUBSCRIBE

Content-Length: 0

 

 

[5] 20110616172811: SIP Tx udp:192.168.1.12:5060:

BYE sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8a685730b04cc2548150ba741c72b373;rport

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

Call-ID: 7e61c163@pbx

CSeq: 19525 BYE

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

Reason: Preemption;cause=3;text="No Media"

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110616172811: SIP Tr udp:192.168.1.12:5060:

BYE sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8a685730b04cc2548150ba741c72b373;rport

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

Call-ID: 7e61c163@pbx

CSeq: 19525 BYE

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

Reason: Preemption;cause=3;text="No Media"

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

 

[5] 20110616172843: Last message repeated 10 times

[7] 20110616172843: Call 7e61c163@pbx: Clear last request

[5] 20110616172843: BYE Response: Terminate 7e61c163@pbx

[5] 20110616172843: SIP Tx tcp:192.168.1.13:65159:

BYE sip:CommServer.creditfree.local:65122;transport=Tcp;maddr=192.168.1.13;ms-opaque=e6d2caedfe360c75 SIP/2.0

Via: SIP/2.0/TCP 192.168.1.13:5060;branch=z9hG4bK-0e77f88bc248991b805201ccdbd1466a;rport

From: <sip:13109644430@192.168.1.13:5060;transport=tcp>;tag=51fdee39e8

To: <sip:0012012181444@192.168.1.12:5060;transport=tcp>;tag=d01782b7fb;epid=834EFDD15A

Call-ID: 9b5e977d-efb2-4bf6-99e7-b07c4e631f0c

CSeq: 21455 BYE

Max-Forwards: 70

Contact: <sip:13109644430@192.168.1.13:5060;transport=tcp>

Content-Length: 0

 

 

[5] 20110616172843: SIP Rx tcp:192.168.1.13:65159:

SIP/2.0 200 OK

FROM: <sip:13109644430@192.168.1.13:5060;transport=tcp>;tag=51fdee39e8

TO: <sip:0012012181444@192.168.1.12:5060;transport=tcp>;tag=d01782b7fb;epid=834EFDD15A

CSEQ: 21455 BYE

CALL-ID: 9b5e977d-efb2-4bf6-99e7-b07c4e631f0c

VIA: SIP/2.0/TCP 192.168.1.13:5060;branch=z9hG4bK-0e77f88bc248991b805201ccdbd1466a;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

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[5] 20110616172602: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:65160;branch=z9hG4bK-814ba4131a80f3dfbb04a4751690ffe9;rport=65160

Contact: <sip:13109644430@192.168.1.12:5060;user=phone>

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

Call-ID: 7e61c163@pbx

CSeq: 19524 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, INFO, OPTIONS, REFER

Content-Type: application/sdp

Supported: replaces, norefersub

User-Agent: HG4000/1.0

Content-Length: 189

 

v=0

o=HG4000 0 0 IN IP4 192.168.1.12

s=HG4000-Session

c=IN IP4 192.168.1.12

t=0 0

m=audio 4010 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[5] 20110616172602: SIP Tx udp:192.168.1.12:5060:

ACK sip:13109644430@192.168.1.12:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-37e2311adbf79cf72758680717621611;rport

From: <sip:0012012181444@192.168.1.12:5060;transport=tcp;user=phone>;tag=15566

To: <sip:13109644430@192.168.1.12;user=phone>;tag=e267f553

Call-ID: 7e61c163@pbx

CSeq: 19524 ACK

Max-Forwards: 70

Contact: <sip:0012012181444@192.168.1.13:5060;transport=udp>

P-Asserted-Identity: "Forty Two" <sip:42@pbx.company.com;user=phone>

Content-Length: 0

 

I guess this is the problem. The contact header in the 200 Ok (coming on on TCP transport layer) tells the PBX to use UDP, which it does. But the ACK obviously does not make it. Maybe you can ask if they intenionally choose UDP for the ACK. I guess not, and maybe/probably the gateway discards the ACK and that is the reason for the trouble.

 

Could it be that there is a firewall in between? For example, in Windows you can tell the personal firewall that UDP is not allowed on a application; maybe for the PBX that is the case. Or if you have a firewall in the network, maybe UDP as not enabled.

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I again tried to set HG4000 Transport = UDP. I also changed the Hypermedia Trunk in PBX to

sip:192.168.1.12:5060;transport=udp

 

The Call does not connect when I call. It look slike HG4000 i strying to make a call. Please see HG4000 Log below. But I see nothing in PBX (192.68.1.13). Should I configure something differetnly in PBX for UDP to work? I will also post a log of HG 4000 where the incoming and outgoing call are working but transfer doe not take place.

 

Looking at the above observation & as user "snom ONE" suggested, the firewall on the PBX or on the gateway is dropping the UDP SIP packets. Is port 5060 opened for both TCP & UDP protocols?

 

BTW, just FYI, if the logs are attached as a file, it is easier to read the post ;)

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Looking at the above observation & as user "snom ONE" suggested, the firewall on the PBX or on the gateway is dropping the UDP SIP packets. Is port 5060 opened for both TCP & UDP protocols?

 

BTW, just FYI, if the logs are attached as a file, it is easier to read the post ;)

 

Thanks for feedback. I turned off the windows firewall on the server. Now calls get routed from HG4000 to PBX on the UDP protocol. I changed the protocol to UDP on the HG4000 and also the trunk on PBX to state transport=udp. I place a call to HG4000. I gets routed to speech server through PBX. I answer the IVR Prompt. Speech server places teh consulation call through PBX. I receive the call. Speech server plays the IVR prompt. I enter the code. So far so good. As soon as I enter the code, My application does a transfer (asyncTransfer) and both th ephones hang up immediately.

 

PBX log shows timeout. The FULL Log is attached. following is the section that shows timeout. 192.168.1.13:6060 is my trunk to speech server

 

 

[6] 20110617102002: SIP TCP/TLS timeout on 192.168.1.13:6060, closing connection

[5] 20110617102007: SIP Rx tcp:192.168.1.13:49341:

REFER sip:0012012181411@192.168.1.13:49413;transport=tcp SIP/2.0

FROM: <sip:8768776075@192.168.1.13:6060;user=phone>;epid=84769DFAC6;tag=41d59776aa

TO: <sip:0012012181411@192.168.1.12:5060;user=phone>;tag=44336

CSEQ: 2 REFER

CALL-ID: 8e255b39@pbx

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.1.13:49341;branch=z9hG4bK96a9afa1

CONTACT: <sip:CommServer.creditfree.local:49341;transport=Tcp;maddr=192.168.1.13;ms-opaque=8e287a1fdc510a6b>;automata

CONTENT-LENGTH: 0

REFER-TO: <sip:12012181444@192.168.1.13:49413;transport=tcp;user=phone?REPLACES=0c17f648-a2b6-46ed-85a5-c1c0cdcbd336%3Bto-tag%3Ddaa53a4d54%3Bfrom-tag%3D9353450d8>

REFERRED-BY: <sip:8768776075@192.168.1.13:6060;user=phone>

USER-AGENT: RTCC/3.0.0.0

Log.txt

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The issue is this - [7] 20110617102007: REFER from device type "RTCC/3.0.0.0" is not supported in this product.

 

snomONE blue/yellow/free editions do not support REFER from 3rd party devices. We will inform the marketing / licensing group to see if they have any solution/workaround.

 

can you please let me know as soon as possible? Also, can you please let me know if this works on snomONE Plus Edition. If Yes, what is the cost of this device for ~150 extensions.

 

Thanks,

 

Sanjeev.

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