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voicemail times out


joso
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i have the audiocodes mp-118 fxo connected to pbxnsip and ocs 2007 and exchange 2007. When i call in, the voicemail kicks in with the exchange attendant. after the beep it seems it always disconnects after 10sec. Am i missing a setting here?

 

Could be a problem with the ACK routing (do you have more than one IP address or a firewall somewhere?). Or could be a tone detection that is too aggressive detecting a hangup.

 

Maybe you can get the SIP packets between the AC and the PBX and we can see if the hangup comes from there. If that does not give any insight, we can do the same thing between the PBX and Exchange. Divide and conquer.

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I only have one firewall and ip at the moment. I have the log file, not sure if it will help:

 

Also, what does this mean "SIP TCP/TLS timeout on 192.168.1.2:42354, closing connection "

 

5] 2008/09/18 15:11:12: SIP port accept from 192.168.1.2:42393

[7] 2008/09/18 15:11:21: SIP Rx udp:192.168.1.6:5060:

INVITE sip:301@192.168.1.5;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2019376203

Max-Forwards: 70

From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919

To: <sip:301@192.168.1.5;user=phone>

Call-ID: 201937152741200005929@192.168.1.6

CSeq: 1 INVITE

Contact: <sip:4032708885@192.168.1.6>

Supported: em,100rel,timer,replaces,path,early-session,resource-priority

Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE

User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.20A.027.004

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 287

 

v=0

o=AudiocodesGW 2019365073 2019364953 IN IP4 192.168.1.6

s=Phone-Call

c=IN IP4 192.168.1.6

t=0 0

m=audio 6000 RTP/AVP 8 0 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

a=rtcp:6001 IN IP4 192.168.1.6

 

[7] 2008/09/18 15:11:21: UDP: Opening socket on port 50166

[7] 2008/09/18 15:11:21: UDP: Opening socket on port 50167

[5] 2008/09/18 15:11:21: Identify trunk (domain name match) 2

[7] 2008/09/18 15:11:21: SIP Tx udp:192.168.1.6:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2019376203

From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919

To: <sip:301@192.168.1.5;user=phone>;tag=ebaf4451bd

Call-ID: 201937152741200005929@192.168.1.6

CSeq: 1 INVITE

Content-Length: 0

 

 

[7] 2008/09/18 15:11:21: Set packet length to 20

[6] 2008/09/18 15:11:21: Sending RTP for 201937152741200005929@192.168.1.6#ebaf4451bd to 192.168.1.6:6000

[5] 2008/09/18 15:11:21: Trunk AudioCodes sends call to 301

[7] 2008/09/18 15:11:21: Set packet length to 20

[7] 2008/09/18 15:11:21: SIP Tx udp:192.168.1.6:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2019376203

From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919

To: <sip:301@192.168.1.5;user=phone>;tag=ebaf4451bd

Call-ID: 201937152741200005929@192.168.1.6

CSeq: 1 INVITE

Contact: <sip:josog@192.168.1.5:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2998

Content-Type: application/sdp

Content-Length: 226

 

v=0

o=- 62207 62207 IN IP4 192.168.1.5

s=-

c=IN IP4 192.168.1.5

t=0 0

m=audio 50166 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=sendrecv

 

[7] 2008/09/18 15:11:21: Last message repeated 2 times

[7] 2008/09/18 15:11:21: SIP Rx udp:192.168.1.6:5060:

ACK sip:josog@192.168.1.5:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2019562213

Max-Forwards: 70

From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919

To: <sip:301@192.168.1.5;user=phone>;tag=ebaf4451bd

Call-ID: 201937152741200005929@192.168.1.6

CSeq: 1 ACK

Contact: <sip:4032708885@192.168.1.6>

Supported: em,timer,replaces,path,early-session,resource-priority

Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE

User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.20A.027.004

Content-Length: 0

 

 

[7] 2008/09/18 15:11:29: Last message repeated 2 times

[6] 2008/09/18 15:11:29: Received DTMF 3

[6] 2008/09/18 15:11:29: Received DTMF 2

[6] 2008/09/18 15:11:29: Received DTMF 5

[7] 2008/09/18 15:11:32: Calling extension 325

[7] 2008/09/18 15:11:32: UDP: Opening socket on port 60722

[7] 2008/09/18 15:11:32: UDP: Opening socket on port 60723

[5] 2008/09/18 15:11:32: Dialplan jngconsulting: Match 4037105450@localhost to <sip:4037105450@192.168.1.6;user=phone> on trunk AudioCodes

[5] 2008/09/18 15:11:32: Using "Subnet" <sip:4032708885@localhost;user=phone> as redirect from

[5] 2008/09/18 15:11:32: Charge user 325 for redirecting calls

[7] 2008/09/18 15:11:32: SIP Tx tcp:192.168.1.4:5060:

INVITE sip:+4036707140325@ssi-ocsmed.jngconsulting.com;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport

From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=4673

To: "Joso Grivicic" <sip:325@localhost>

Call-ID: 82331c2d@pbx

CSeq: 12820 INVITE

Max-Forwards: 70

Contact: <sip:325@192.168.1.5:4638;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2998

Alert-Info: <http://127.0.0.1/Bellcore-dr3>

Content-Type: application/sdp

Content-Length: 335

 

v=0

o=- 60462 60462 IN IP4 192.168.1.5

s=-

c=IN IP4 192.168.1.5

t=0 0

m=audio 60722 RTP/AVP 0 8 9 18 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/09/18 15:11:32: UDP: Opening socket on port 55700

[7] 2008/09/18 15:11:32: UDP: Opening socket on port 55701

[7] 2008/09/18 15:11:32: SIP Tx udp:192.168.1.22:5060:

INVITE sip:325@192.168.1.22 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport

From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=62862

To: "Joso Grivicic" <sip:325@localhost>

Call-ID: b016bc51@pbx

CSeq: 4524 INVITE

Max-Forwards: 70

Contact: <sip:325@192.168.1.5:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2998

Alert-Info: <http://127.0.0.1/Bellcore-dr3>

Content-Type: application/sdp

Content-Length: 335

 

v=0

o=- 23309 23309 IN IP4 192.168.1.5

s=-

c=IN IP4 192.168.1.5

t=0 0

m=audio 55700 RTP/AVP 0 8 9 18 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/09/18 15:11:32: UDP: Opening socket on port 51064

[7] 2008/09/18 15:11:32: UDP: Opening socket on port 51065

[7] 2008/09/18 15:11:32: SIP Tx udp:192.168.1.6:5060:

INVITE sip:4037105450@192.168.1.6;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-1faedd9e7819005bb387250f371f51f2;rport

From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=65392

To: <sip:4037105450@192.168.1.6;user=phone>

Call-ID: 5f33b786@pbx

CSeq: 14343 INVITE

Max-Forwards: 70

Contact: <sip:josog@192.168.1.5:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2998

P-Asserted-Identity: <sip:josog@192.168.1.6;user=phone>

Content-Type: application/sdp

Content-Length: 335

 

v=0

o=- 44121 44121 IN IP4 192.168.1.5

s=-

c=IN IP4 192.168.1.5

t=0 0

m=audio 51064 RTP/AVP 0 8 9 18 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/09/18 15:11:32: SIP Rx tcp:192.168.1.4:5060:

SIP/2.0 100 Trying

FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=4673

TO: "Joso Grivicic"<sip:325@localhost>

CSEQ: 12820 INVITE

CALL-ID: 82331c2d@pbx

VIA: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport

CONTENT-LENGTH: 0

 

 

[7] 2008/09/18 15:11:32: SIP Rx udp:192.168.1.6:5060:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-1faedd9e7819005bb387250f371f51f2;rport

From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=65392

To: <sip:4037105450@192.168.1.6;user=phone>;tag=1c2046102598

Call-ID: 5f33b786@pbx

CSeq: 14343 INVITE

Supported: em,timer,replaces,path,resource-priority

Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE

Server: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.20A.027.004

Reason: Q.850 ;cause=3

Content-Length: 0

 

 

[7] 2008/09/18 15:11:32: Call 5f33b786@pbx#65392: Clear last INVITE

[7] 2008/09/18 15:11:32: SIP Tx udp:192.168.1.6:5060:

ACK sip:4037105450@192.168.1.6;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-1faedd9e7819005bb387250f371f51f2;rport

From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=65392

To: <sip:4037105450@192.168.1.6;user=phone>;tag=1c2046102598

Call-ID: 5f33b786@pbx

CSeq: 14343 ACK

Max-Forwards: 70

Contact: <sip:josog@192.168.1.5:5060;transport=udp>

P-Asserted-Identity: <sip:josog@192.168.1.6;user=phone>

Content-Length: 0

 

 

[5] 2008/09/18 15:11:32: INVITE Response: Terminate 5f33b786@pbx

[7] 2008/09/18 15:11:32: Other Ports: 3

[7] 2008/09/18 15:11:32: Call Port: 201937152741200005929@192.168.1.6#ebaf4451bd

[7] 2008/09/18 15:11:32: Call Port: 82331c2d@pbx#4673

[7] 2008/09/18 15:11:32: Call Port: b016bc51@pbx#62862

[7] 2008/09/18 15:11:32: SIP Rx udp:192.168.1.22:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport

From:"Subnet" <sip:4032708885@localhost;user=phone>;tag=62862

To:"Joso Grivicic" <sip:325@localhost>;tag=48d2379a-220-6359b15a

CSeq:4524 INVITE

User-Agent:Mitel-5220-SIP-Phone 03.00.00.01 08000F178504

Call-ID:b016bc51@pbx

Content-Length:0

 

 

[7] 2008/09/18 15:11:32: SIP Rx udp:192.168.1.22:5060:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport

From:"Subnet" <sip:4032708885@localhost;user=phone>;tag=62862

To:"Joso Grivicic" <sip:325@localhost>;tag=48d2379a-220-6359b15a

CSeq:4524 INVITE

User-Agent:Mitel-5220-SIP-Phone 03.00.00.01 08000F178504

Call-ID:b016bc51@pbx

Content-Length:0

 

 

[7] 2008/09/18 15:11:32: SIP Rx tcp:192.168.1.4:5060:

SIP/2.0 183 Session Progress

FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=4673

TO: Joso Grivicic<sip:325@localhost>;epid=0FDC8A2A83;tag=49601a2ec

CSEQ: 12820 INVITE

CALL-ID: 82331c2d@pbx

VIA: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0 MediationServer

 

 

[7] 2008/09/18 15:11:32: SIP Rx tcp:192.168.1.4:5060:

SIP/2.0 180 Ringing

FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=4673

TO: Joso Grivicic<sip:325@localhost>;epid=0FDC8A2A83;tag=49601a2ec

CSEQ: 12820 INVITE

CALL-ID: 82331c2d@pbx

VIA: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0 MediationServer

 

 

[7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.4:5060:

CANCEL sip:+4036707140325@ssi-ocsmed.jngconsulting.com;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport

From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=4673

To: "Joso Grivicic" <sip:325@localhost>

Call-ID: 82331c2d@pbx

CSeq: 12820 CANCEL

Max-Forwards: 70

Content-Length: 0

 

 

[6] 2008/09/18 15:11:42: Redirecting to external voicemail account 325 destination sip:7325@localhost

[5] 2008/09/18 15:11:42: Dialplan jngconsulting: Match 7325@localhost to <sip:325@192.168.1.2;user=phone> on trunk Exchange

[5] 2008/09/18 15:11:42: Using "Subnet" <sip:4032708885@localhost;user=phone> as redirect from

[7] 2008/09/18 15:11:42: SIP Tx udp:192.168.1.22:5060:

CANCEL sip:325@192.168.1.22 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport

From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=62862

To: "Joso Grivicic" <sip:325@localhost>

Call-ID: b016bc51@pbx

CSeq: 4524 CANCEL

Max-Forwards: 70

Content-Length: 0

 

 

[5] 2008/09/18 15:11:42: Charge user 325 for redirecting calls

[7] 2008/09/18 15:11:42: UDP: Opening socket on port 57840

[7] 2008/09/18 15:11:42: UDP: Opening socket on port 57841

[7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.2:5060:

INVITE sip:325@192.168.1.2;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.5:4649;branch=z9hG4bK-abcbde6b048db506b99c96a6ac260f49;rport

From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=54625

To: <sip:325@192.168.1.2;user=phone>

Call-ID: 26ed22a4@pbx

CSeq: 19637 INVITE

Max-Forwards: 70

Contact: <sip:4032708885@192.168.1.5:4649;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2998

Diversion: <tel:325>;reason=no-answer;screen=no;privacy=off

P-Asserted-Identity: "Joso Grivicic" <sip:325@192.168.1.2;user=phone>

Content-Type: application/sdp

Content-Length: 335

 

v=0

o=- 11803 11803 IN IP4 192.168.1.5

s=-

c=IN IP4 192.168.1.5

t=0 0

m=audio 57840 RTP/AVP 0 8 9 18 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.2:5060:

SIP/2.0 100 Trying

FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625

TO: <sip:325@192.168.1.2;user=phone>

CSEQ: 19637 INVITE

CALL-ID: 26ed22a4@pbx

VIA: SIP/2.0/TCP 192.168.1.5:4649;branch=z9hG4bK-abcbde6b048db506b99c96a6ac260f49;rport

CONTENT-LENGTH: 0

 

 

[7] 2008/09/18 15:11:42: SIP Rx udp:192.168.1.22:5060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport

From:"Subnet" <sip:4032708885@localhost;user=phone>;tag=62862

To:"Joso Grivicic" <sip:325@localhost>

CSeq:4524 CANCEL

User-Agent:Mitel-5220-SIP-Phone 03.00.00.01 08000F178504

Call-ID:b016bc51@pbx

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

Content-Length:0

 

 

[7] 2008/09/18 15:11:42: Call b016bc51@pbx#62862: Clear last request

[7] 2008/09/18 15:11:42: SIP Rx udp:192.168.1.22:5060:

SIP/2.0 487 Request Cancelled

Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport

From:"Subnet" <sip:4032708885@localhost;user=phone>;tag=62862

To:"Joso Grivicic" <sip:325@localhost>;tag=48d2379a-220-6359b15a

CSeq:4524 INVITE

User-Agent:Mitel-5220-SIP-Phone 03.00.00.01 08000F178504

Call-ID:b016bc51@pbx

Content-Length:0

 

 

[7] 2008/09/18 15:11:42: Call b016bc51@pbx#62862: Clear last INVITE

[7] 2008/09/18 15:11:42: SIP Tx udp:192.168.1.22:5060:

ACK sip:325@192.168.1.22 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK-09918fb1a67eb8dd875c7cbbf513f878;rport

From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=62862

To: "Joso Grivicic" <sip:325@localhost>;tag=48d2379a-220-6359b15a

Call-ID: b016bc51@pbx

CSeq: 4524 ACK

Max-Forwards: 70

Contact: <sip:325@192.168.1.5:5060;transport=udp>

Content-Length: 0

 

 

[5] 2008/09/18 15:11:42: INVITE Response: Terminate b016bc51@pbx

[7] 2008/09/18 15:11:42: Other Ports: 3

[7] 2008/09/18 15:11:42: Call Port: 201937152741200005929@192.168.1.6#ebaf4451bd

[7] 2008/09/18 15:11:42: Call Port: 26ed22a4@pbx#54625

[7] 2008/09/18 15:11:42: Call Port: 82331c2d@pbx#4673

[7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.4:5060:

SIP/2.0 487 Request Terminated

FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=4673

TO: Joso Grivicic<sip:325@localhost>;epid=0FDC8A2A83;tag=49601a2ec

CSEQ: 12820 INVITE

CALL-ID: 82331c2d@pbx

VIA: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0 MediationServer

 

 

[7] 2008/09/18 15:11:42: Call 82331c2d@pbx#4673: Clear last INVITE

[7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.4:5060:

ACK sip:+4036707140325@ssi-ocsmed.jngconsulting.com;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport

From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=4673

To: "Joso Grivicic" <sip:325@localhost>;tag=49601a2ec

Call-ID: 82331c2d@pbx

CSeq: 12820 ACK

Max-Forwards: 70

Contact: <sip:325@192.168.1.5:4638;transport=tcp>

Content-Length: 0

 

 

[5] 2008/09/18 15:11:42: INVITE Response: Terminate 82331c2d@pbx

[7] 2008/09/18 15:11:42: Other Ports: 2

[7] 2008/09/18 15:11:42: Call Port: 201937152741200005929@192.168.1.6#ebaf4451bd

[7] 2008/09/18 15:11:42: Call Port: 26ed22a4@pbx#54625

[7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.4:5060:

SIP/2.0 200 OK

FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=4673

TO: Joso Grivicic<sip:325@localhost>;epid=0FDC8A2A83;tag=bd82658dfc

CSEQ: 12820 CANCEL

CALL-ID: 82331c2d@pbx

VIA: SIP/2.0/TCP 192.168.1.5:4638;branch=z9hG4bK-433167005897384ba08f53293f0ca7ca;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0 MediationServer

 

 

[7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.2:5060:

SIP/2.0 302 Moved Temporarily

FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625

TO: <sip:325@192.168.1.2;user=phone>;tag=26226ab74c

CSEQ: 19637 INVITE

CALL-ID: 26ed22a4@pbx

VIA: SIP/2.0/TCP 192.168.1.5:4649;branch=z9hG4bK-abcbde6b048db506b99c96a6ac260f49;rport

CONTACT: <sip:325@192.168.1.2:5065;user=phone;transport=TCP>

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

Diversion: <tel:325>;reason=no-answer;screen=no;privacy=off

 

 

[7] 2008/09/18 15:11:42: Call 26ed22a4@pbx#54625: Clear last INVITE

[7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.2:5060:

ACK sip:325@192.168.1.2;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.5:4649;branch=z9hG4bK-abcbde6b048db506b99c96a6ac260f49;rport

From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=54625

To: <sip:325@192.168.1.2;user=phone>;tag=26226ab74c

Call-ID: 26ed22a4@pbx

CSeq: 19637 ACK

Max-Forwards: 70

Contact: <sip:4032708885@192.168.1.5:4649;transport=tcp>

P-Asserted-Identity: "Joso Grivicic" <sip:325@192.168.1.2;user=phone>

Content-Length: 0

 

 

[5] 2008/09/18 15:11:42: Redirecting call

[7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.2:5065:

INVITE sip:325@192.168.1.2:5065;user=phone;transport=TCP SIP/2.0

Via: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-9ba3285162ecc8be7a001b93c1904328;rport

From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=54625

To: <sip:325@192.168.1.2;user=phone>

Call-ID: 26ed22a4@pbx

CSeq: 19638 INVITE

Max-Forwards: 70

Contact: <sip:4032708885@192.168.1.5:4650;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2998

Diversion: <tel:325>;reason=no-answer;screen=no;privacy=off

P-Asserted-Identity: "Joso Grivicic" <sip:325@192.168.1.2;user=phone>

Content-Type: application/sdp

Content-Length: 335

 

v=0

o=- 11803 11803 IN IP4 192.168.1.5

s=-

c=IN IP4 192.168.1.5

t=0 0

m=audio 57840 RTP/AVP 0 8 9 18 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.2:5065:

SIP/2.0 100 Trying

FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625

TO: <sip:325@192.168.1.2;user=phone>

CSEQ: 19638 INVITE

CALL-ID: 26ed22a4@pbx

VIA: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-9ba3285162ecc8be7a001b93c1904328;rport

CONTENT-LENGTH: 0

 

 

[7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.2:5065:

SIP/2.0 180 Ringing

FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625

TO: <sip:325@192.168.1.2;user=phone>;epid=7EF0970BA2;tag=6b14a423e3

CSEQ: 19638 INVITE

CALL-ID: 26ed22a4@pbx

VIA: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-9ba3285162ecc8be7a001b93c1904328;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[7] 2008/09/18 15:11:42: SIP Rx tcp:192.168.1.2:5065:

SIP/2.0 200 OK

FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625

TO: <sip:325@192.168.1.2;user=phone>;epid=7EF0970BA2;tag=6b14a423e3

CSEQ: 19638 INVITE

CALL-ID: 26ed22a4@pbx

VIA: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-9ba3285162ecc8be7a001b93c1904328;rport

CONTACT: <sip:SSI-TESTDC.jngconsulting.com:5065;transport=Tcp;maddr=192.168.1.2>;automata

CONTENT-LENGTH: 192

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 192.168.1.2

s=Microsoft Exchange Speech Engine

c=IN IP4 192.168.1.2

t=0 0

m=audio 6272 RTP/AVP 0 8 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

 

[7] 2008/09/18 15:11:42: Call 26ed22a4@pbx#54625: Clear last INVITE

[7] 2008/09/18 15:11:42: Set packet length to 20

[6] 2008/09/18 15:11:42: Sending RTP for 26ed22a4@pbx#54625 to 192.168.1.2:6272

[7] 2008/09/18 15:11:42: SIP Tx tcp:192.168.1.2:5065:

ACK sip:SSI-TESTDC.jngconsulting.com:5065;transport=Tcp;maddr=192.168.1.2 SIP/2.0

Via: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-9869b4983d009ddb2438347e95d0ccb0;rport

From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=54625

To: <sip:325@192.168.1.2;user=phone>;tag=6b14a423e3

Call-ID: 26ed22a4@pbx

CSeq: 19638 ACK

Max-Forwards: 70

Contact: <sip:4032708885@192.168.1.5:4650;transport=tcp>

P-Asserted-Identity: "Joso Grivicic" <sip:325@192.168.1.2;user=phone>

Content-Length: 0

 

 

[7] 2008/09/18 15:11:42: Determine pass-through mode after receiving response

[7] 2008/09/18 15:11:42: 26ed22a4@pbx#54625: RTP pass-through mode

[7] 2008/09/18 15:11:42: 201937152741200005929@192.168.1.6#ebaf4451bd: RTP pass-through mode

[7] 2008/09/18 15:12:02: SIP Rx udp:192.168.1.6:5060:

BYE sip:josog@192.168.1.5:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2122852159

Max-Forwards: 70

From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919

To: <sip:301@192.168.1.5;user=phone>;tag=ebaf4451bd

Call-ID: 201937152741200005929@192.168.1.6

CSeq: 2 BYE

Supported: em,timer,replaces,path,early-session,resource-priority

Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE

User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.20A.027.004

Reason: Q.850 ;cause=31 ;text="RTP Broken Connection"

Content-Length: 0

 

 

[7] 2008/09/18 15:12:02: SIP Tx udp:192.168.1.6:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.1.6;branch=z9hG4bKac2122852159

From: "Subnet" <sip:4032708885@192.168.1.6>;tag=1c2019371919

To: <sip:301@192.168.1.5;user=phone>;tag=ebaf4451bd

Call-ID: 201937152741200005929@192.168.1.6

CSeq: 2 BYE

Contact: <sip:josog@192.168.1.5:5060;transport=udp>

User-Agent: pbxnsip-PBX/3.0.0.2998

RTP-RxStat: Dur=41,Pkt=2064,Oct=350952,Underun=0

RTP-TxStat: Dur=41,Pkt=1545,Oct=265740

Content-Length: 0

 

 

[7] 2008/09/18 15:12:02: 26ed22a4@pbx#54625: Media-aware pass-through mode

[7] 2008/09/18 15:12:02: Other Ports: 1

[7] 2008/09/18 15:12:02: Call Port: 26ed22a4@pbx#54625

[7] 2008/09/18 15:12:02: SIP Tx tcp:192.168.1.2:5065:

BYE sip:SSI-TESTDC.jngconsulting.com:5065;transport=Tcp;maddr=192.168.1.2 SIP/2.0

Via: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-b02d89319eec09d9258589db2e1c8e79;rport

From: "Subnet" <sip:4032708885@localhost;user=phone>;tag=54625

To: <sip:325@192.168.1.2;user=phone>;tag=6b14a423e3

Call-ID: 26ed22a4@pbx

CSeq: 19639 BYE

Max-Forwards: 70

Contact: <sip:4032708885@192.168.1.5:4650;transport=tcp>

RTP-RxStat: Dur=21,Pkt=495,Oct=85140,Underun=0

RTP-TxStat: Dur=20,Pkt=1017,Oct=174924

P-Asserted-Identity: "Joso Grivicic" <sip:325@192.168.1.2;user=phone>

Content-Length: 0

 

 

[7] 2008/09/18 15:12:02: SIP Rx tcp:192.168.1.2:5065:

SIP/2.0 200 OK

FROM: "Subnet"<sip:4032708885@localhost;user=phone>;tag=54625

TO: <sip:325@192.168.1.2;user=phone>;tag=6b14a423e3;epid=7EF0970BA2

CSEQ: 19639 BYE

CALL-ID: 26ed22a4@pbx

VIA: SIP/2.0/TCP 192.168.1.5:4650;branch=z9hG4bK-b02d89319eec09d9258589db2e1c8e79;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

[7] 2008/09/18 15:12:02: Call 26ed22a4@pbx#54625: Clear last request

[5] 2008/09/18 15:12:02: BYE Response: Terminate 26ed22a4@pbx

[6] 2008/09/18 15:12:11: SIP TCP/TLS timeout on 192.168.1.2:42354, closing connection

[5] 2008/09/18 15:13:12: SIP port accept from 192.168.1.2:30667

[6] 2008/09/18 15:14:11: SIP TCP/TLS timeout on 192.168.1.2:42365, closing connection

[5] 2008/09/18 15:15:12: SIP port accept from 192.168.1.2:30674

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I only have one firewall and ip at the moment. I have the log file, not sure if it will help:

 

Also, what does this mean "SIP TCP/TLS timeout on 192.168.1.2:42354, closing connection "

 

Well, that timeout happens after the call is already over. The gateway initiates the hangup, and it says it does that because of RTP problems. You can see in the BYE response that the PBX sends far less packets than it receives. So it is understandable why the gateway hangs up. Maybe you have silence suppression turned on?

 

With media related problems, it makes sense to get a Wireshark trace. Then we will be able to see why the PBX does not send RTP data. Maybe it is because of the Speech Engine, maybe there is a option to disable silence suppression. Also, check if there is a upgrade availale. 3.0.0 sounds like there is...

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I think i found the problem. In the audiocodes mp-118 under coders. I had it set at G.711U-Law, when i changed this to G.711A-Law it works fine. Silence SUpression has always been disabled.

 

What setting should this be set at, since there are many to choose from.

 

Joso

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I think i found the problem. In the audiocodes mp-118 under coders. I had it set at G.711U-Law, when i changed this to G.711A-Law it works fine. Silence SUpression has always been disabled.

 

What setting should this be set at, since there are many to choose from.

 

Whow, that means the AC has a problem with the codec negotiation? The PBX answers with it's own priority (which is obviously Ulaw, then Alaw), and maybe the AC cannot deal with that... Anyway, if it works then keep it this way.

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