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speech server 2007 - snom on - outbound call - authenticatin isue


callsanjeevat
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I am trying to place an outgoing call to PSTN from Microsoft Speech server 2007 through SNOM One PBX. But SNOM One is asking for authentication

 

 

My set up is as follows

 

1. speech server - Created a trusted peer to snom one pbx at localhost:5060 (speech server is listening on 6060)

2. pbxnsip - created a trunk (type - sip proxy) to speech server - Assume call comes from - extension 42.

3. PBXnSIP - created a trunk (type - sip gateway) to Hypermedia Gateway.

4. PBXnSIP - Created a dialplan "Hypermedia-Dialplan" and set Trunk for the dial plan to Hypermedia (trunk created above)

5. PBXnSIP - For extension 42 , dial plan is set to hyprmedia-dialplan

6. When I place an outgoing call from speech server, I can see that the sip request is reachig the hypermedia gateway. But Speech server is complaining as follows

 

An error occurred during call transfer: Microsoft.SpeechServer.SipPeerException: A SIP request has failed. The current operation is 'Opening'. The session state is 'Connecting'. The remote participant is 'sip:2012181444@192.168.1.13:5060;user=phone'. The response code was '401'. The response text was 'Authentication Required'. ---> SupportedAuthenticationProtocols=None

Realm=

FailureReason=None

ErrorCode=0

ResponseCode=401 ResponseText=Authentication Required

Microsoft.Rtc.Signaling.AuthenticationException: Peer to peer endpoint does not support authentication.

at Microsoft.Rtc.Signaling.SipAsyncResult.ThrowIfFailed()

at Microsoft.Rtc.Signaling.Helper.EndAsyncOperation[T](Object owner, IAsyncResult asyncResult)

at Microsoft.Rtc.Signaling.Helper.EndAsyncOperation[T](Object owner, IAsyncResult asyncResult, String operationId)

at Microsoft.Rtc.Signaling.SignalingSession.EndEnter(SipInviteAsyncResultWrapper asyncWrapper)

at Microsoft.SpeechServer.Core.TelephonySessionOutbound.ParticipateCallback(IAsyncResult result)

 

5. In Snomone log I see the following...

 

 

[8] 2011/06/10 10:38:51: Received SIP connection 255 from 192.168.1.13:53040

[9] 2011/06/10 10:38:51: SIP Rx tcp:192.168.1.13:53040:

INVITE sip:2012181444@192.168.1.13:5060;user=phone SIP/2.0

FROM: <sip:45@CommServer.creditfree.local:50681;user=phone>;epid=457BE1388F;tag=8b9960a8ed

TO: <sip:2012181444@192.168.1.13:5060;user=phone>

CSEQ: 1 INVITE

CALL-ID: 79644d90-3c04-4839-98a7-729c9b9a6b93

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.1.13:53040;branch=z9hG4bK641e3c71

CONTACT: <sip:CommServer.creditfree.local:50681;transport=Tcp;maddr=192.168.1.13;ms-opaque=b2dfee4fee6e4834>;automata

CONTENT-LENGTH: 335

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 6274 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

 

[9] 2011/06/10 10:38:51: SIP Tx tcp:192.168.1.13:53040:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 192.168.1.13:53040;branch=z9hG4bK641e3c71

From: <sip:45@CommServer.creditfree.local:50681;user=phone>;tag=8b9960a8ed;epid=457BE1388F

To: <sip:2012181444@192.168.1.13:5060;user=phone>;tag=ea32e7f2fd

Call-ID: 79644d90-3c04-4839-98a7-729c9b9a6b93

CSeq: 1 INVITE

Content-Length: 0

 

 

[9] 2011/06/10 10:38:51: SIP Tx tcp:192.168.1.13:53040:

SIP/2.0 401 Authentication RequiredVia: SIP/2.0/TCP 192.168.1.13:53040;branch=z9hG4bK641e3c71

From: <sip:45@CommServer.creditfree.local:50681;user=phone>;tag=8b9960a8ed;epid=457BE1388F

To: <sip:2012181444@192.168.1.13:5060;user=phone>;tag=ea32e7f2fd

Call-ID: 79644d90-3c04-4839-98a7-729c9b9a6b93

CSeq: 1 INVITE

User-Agent: snom-PBX/4.2.0.3950

WWW-Authenticate: Digest realm="commserver.creditfree.local",nonce="d2e7e6915156a03a81283494f153136a",domain="sip:2012181444@192.168.1.13:5060;user=phone",algorithm=MD5

Content-Length: 0

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I Just blanked the Password for Extension 45 and the Authentication problem went away. however, Now I have a different problem. When I place an outgoing call, the phone rings but I cannot hear the Speech server prompt. It is silent. I have included the partial log that shows sections where I think the problem lies. Also The etire log is provided. As soon as I trigger the call from speech server, I check teh log and it shows "ession in progress". The phone rings and I pick up the call. I hear nothing. In the log the TCP/TLS connection times out... and later the transport changes to UDP and it is no longer tcp. It then complains that "[8] 2011/06/10 17:20:21: Call 37a2fda3@pbx: Response does not correspond to open request "

 

any idea how to resolve this? BTW My gateway Hypermedia HG4000.

 

Partial Log[9] 2011/06/10 17:20:06: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TCP 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport=50230

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

To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

From: "Forty Five"<sip:45@pbx.company.com;user=phone>;tag=31523

Call-ID: 37a2fda3@pbx

CSeq: 27729 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 4000 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[6] 2011/06/10 17:20:14: SIP TCP/TLS timeout on 192.168.1.13:50228, closing connection

[8] 2011/06/10 17:20:14: Release SIP thread 324

[9] 2011/06/10 17:20:21: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport=50230

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

To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

From: "Forty Five"<sip:45@pbx.company.com;user=phone>;tag=31523

Call-ID: 37a2fda3@pbx

CSeq: 27729 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 4000 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[7] 2011/06/10 17:20:21: Call 37a2fda3@pbx: Clear last INVITE

[9] 2011/06/10 17:20:21: Resolve 902: url sip:12012181444@192.168.1.12:5060;user=phone

[9] 2011/06/10 17:20:21: Resolve 902: udp 192.168.1.12 5060

[9] 2011/06/10 17:20:21: 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-ca096733c6c524b041e5e2755d954e74;rport

From: "Forty Five" <sip:45@pbx.company.com;user=phone>;tag=31523

To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

Call-ID: 37a2fda3@pbx

CSeq: 27729 ACK

Max-Forwards: 70

Contact: <sip:45@192.168.1.13:5060;transport=udp>

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

Content-Length: 0

 

 

[9] 2011/06/10 17:20:21: Resolve 903: tcp 192.168.1.13 50228

[6] 2011/06/10 17:20:21: Response to 192.168.1.13:50228 must be sent over existing connection

[9] 2011/06/10 17:20:21: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport=50230

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

To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

From: "Forty Five"<sip:45@pbx.company.com;user=phone>;tag=31523

Call-ID: 37a2fda3@pbx

CSeq: 27729 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 4000 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[8] 2011/06/10 17:20:21: Call 37a2fda3@pbx: Response does not correspond to open request

[9] 2011/06/10 17:20:21: Resolve 904: url sip:12012181444@192.168.1.12:5060;user=phone

[9] 2011/06/10 17:20:21: Resolve 904: udp 192.168.1.12 5060

[9] 2011/06/10 17:20:21: 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-ca096733c6c524b041e5e2755d954e74;rport

From: "Forty Five" <sip:45@pbx.company.com;user=phone>;tag=31523

To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

Call-ID: 37a2fda3@pbx

CSeq: 27729 ACK

Max-Forwards: 70

Contact: <sip:45@192.168.1.13:5060;transport=udp>

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

Content-Length: 0

 

 

 

 

 

[8] 2011/06/10 17:20:06: Received SIP connection 324 from 192.168.1.13:50228

[9] 2011/06/10 17:20:06: SIP Rx tcp:192.168.1.13:50228:

INVITE sip:12012181444@192.168.1.13:5060;user=phone SIP/2.0

FROM: <sip:45@CommServer.creditfree.local:50682;user=phone>;epid=67717EB21D;tag=9751c85997

TO: <sip:12012181444@192.168.1.13:5060;user=phone>

CSEQ: 9 INVITE

CALL-ID: b8384679-1360-46f3-9c8a-74235a9f71f9

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 192.168.1.13:50228;branch=z9hG4bK202785c

CONTACT: <sip:CommServer.creditfree.local:50682;transport=Tcp;maddr=192.168.1.13;ms-opaque=f2f2b3b4e0c93efb>;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 64000 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

 

[9] 2011/06/10 17:20:06: SIP Tx tcp:192.168.1.13:50228:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 192.168.1.13:50228;branch=z9hG4bK202785c

From: <sip:45@CommServer.creditfree.local:50682;user=phone>;tag=9751c85997;epid=67717EB21D

To: <sip:12012181444@192.168.1.13:5060;user=phone>;tag=6762cade95

Call-ID: b8384679-1360-46f3-9c8a-74235a9f71f9

CSeq: 9 INVITE

Content-Length: 0

 

 

[8] 2011/06/10 17:20:06: Set the To domain based on From user 45@pbx.company.com

[9] 2011/06/10 17:20:06: Resolve 900: url sip:192.168.1.12:5060;transport=tcp

[9] 2011/06/10 17:20:06: Resolve 900: a tcp 192.168.1.12 5060

[9] 2011/06/10 17:20:06: Resolve 900: tcp 192.168.1.12 5060

[8] 2011/06/10 17:20:06: Received SIP connection 325 from 192.168.1.12:5060

[9] 2011/06/10 17:20:06: 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 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport

From: "Forty Five" <sip:45@pbx.company.com;user=phone>;tag=31523

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

Call-ID: 37a2fda3@pbx

CSeq: 27729 INVITE

Max-Forwards: 70

Contact: <sip:45@192.168.1.13:50230;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/4.2.0.3950

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

Content-Type: application/sdp

Content-Length: 327

 

v=0

o=- 63976 63976 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 31730 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

 

[9] 2011/06/10 17:20:06: SIP Tx tcp:192.168.1.13:50228:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TCP 192.168.1.13:50228;branch=z9hG4bK202785c

From: <sip:45@CommServer.creditfree.local:50682;user=phone>;tag=9751c85997;epid=67717EB21D

To: <sip:12012181444@192.168.1.13:5060;user=phone>;tag=6762cade95

Call-ID: b8384679-1360-46f3-9c8a-74235a9f71f9

CSeq: 9 INVITE

Contact: <sip:45@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/4.2.0.3950

Content-Type: application/sdp

Content-Length: 264

 

v=0

o=- 57454 57454 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 6566 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

 

[9] 2011/06/10 17:20:06: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TCP 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport=50230

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

To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

From: "Forty Five"<sip:45@pbx.company.com;user=phone>;tag=31523

Call-ID: 37a2fda3@pbx

CSeq: 27729 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 4000 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[6] 2011/06/10 17:20:14: SIP TCP/TLS timeout on 192.168.1.13:50228, closing connection

[8] 2011/06/10 17:20:14: Release SIP thread 324

[9] 2011/06/10 17:20:21: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport=50230

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

To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

From: "Forty Five"<sip:45@pbx.company.com;user=phone>;tag=31523

Call-ID: 37a2fda3@pbx

CSeq: 27729 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 4000 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[7] 2011/06/10 17:20:21: Call 37a2fda3@pbx: Clear last INVITE

[9] 2011/06/10 17:20:21: Resolve 902: url sip:12012181444@192.168.1.12:5060;user=phone

[9] 2011/06/10 17:20:21: Resolve 902: udp 192.168.1.12 5060

[9] 2011/06/10 17:20:21: 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-ca096733c6c524b041e5e2755d954e74;rport

From: "Forty Five" <sip:45@pbx.company.com;user=phone>;tag=31523

To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

Call-ID: 37a2fda3@pbx

CSeq: 27729 ACK

Max-Forwards: 70

Contact: <sip:45@192.168.1.13:5060;transport=udp>

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

Content-Length: 0

 

 

[9] 2011/06/10 17:20:21: Resolve 903: tcp 192.168.1.13 50228

[6] 2011/06/10 17:20:21: Response to 192.168.1.13:50228 must be sent over existing connection

[9] 2011/06/10 17:20:21: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport=50230

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

To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

From: "Forty Five"<sip:45@pbx.company.com;user=phone>;tag=31523

Call-ID: 37a2fda3@pbx

CSeq: 27729 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 4000 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[8] 2011/06/10 17:20:21: Call 37a2fda3@pbx: Response does not correspond to open request

[9] 2011/06/10 17:20:21: Resolve 904: url sip:12012181444@192.168.1.12:5060;user=phone

[9] 2011/06/10 17:20:21: Resolve 904: udp 192.168.1.12 5060

[9] 2011/06/10 17:20:21: 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-ca096733c6c524b041e5e2755d954e74;rport

From: "Forty Five" <sip:45@pbx.company.com;user=phone>;tag=31523

To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

Call-ID: 37a2fda3@pbx

CSeq: 27729 ACK

Max-Forwards: 70

Contact: <sip:45@192.168.1.13:5060;transport=udp>

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

Content-Length: 0

 

 

[9] 2011/06/10 17:20:22: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport=50230

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

To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

From: "Forty Five"<sip:45@pbx.company.com;user=phone>;tag=31523

Call-ID: 37a2fda3@pbx

CSeq: 27729 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 4000 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[8] 2011/06/10 17:20:22: Call 37a2fda3@pbx: Response does not correspond to open request

[9] 2011/06/10 17:20:22: Resolve 905: url sip:12012181444@192.168.1.12:5060;user=phone

[9] 2011/06/10 17:20:22: Resolve 905: udp 192.168.1.12 5060

[9] 2011/06/10 17:20:22: 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-ca096733c6c524b041e5e2755d954e74;rport

From: "Forty Five" <sip:45@pbx.company.com;user=phone>;tag=31523

To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

Call-ID: 37a2fda3@pbx

CSeq: 27729 ACK

Max-Forwards: 70

Contact: <sip:45@192.168.1.13:5060;transport=udp>

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

Content-Length: 0

 

 

[9] 2011/06/10 17:20:24: SIP Rx tcp:192.168.1.12:5060:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 192.168.1.13:50230;branch=z9hG4bK-ce635a49a2a692a1bfee46cad8bc6e27;rport=50230

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

To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

From: "Forty Five"<sip:45@pbx.company.com;user=phone>;tag=31523

Call-ID: 37a2fda3@pbx

CSeq: 27729 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 4000 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

[8] 2011/06/10 17:20:24: Call 37a2fda3@pbx: Response does not correspond to open request

[9] 2011/06/10 17:20:24: Resolve 906: url sip:12012181444@192.168.1.12:5060;user=phone

[9] 2011/06/10 17:20:24: Resolve 906: udp 192.168.1.12 5060

[9] 2011/06/10 17:20:24: 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-ca096733c6c524b041e5e2755d954e74;rport

From: "Forty Five" <sip:45@pbx.company.com;user=phone>;tag=31523

To: <sip:12012181444@192.168.1.12;user=phone>;tag=4a728d6f

Call-ID: 37a2fda3@pbx

CSeq: 27729 ACK

Max-Forwards: 70

Contact: <sip:45@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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The timeout problem is solved now. I had to change the following on the trunk. Please note that request timeout will be visible only when Failover behaviour default is changed

 

Failover Behavior: except for busy response

Request Timeout: 30

 

Now the Last Leg of my proof-of-concept project... To do a supervised transfer. i will post back afteri am finished.

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A few months ago, we found out that the PBX closes a connection that has not been authenticated with a successful registration, after 8 seconds (this is to keep hackers away). We extended the logic so that also successful INVITE requests also keep the connection alive. It seems like you dont have the latest version (where did you get the link from), maybe just private message pbx_support, indicate your OS and then we'll send you the link with the latest build.

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A few months ago, we found out that the PBX closes a connection that has not been authenticated with a successful registration, after 8 seconds (this is to keep hackers away). We extended the logic so that also successful INVITE requests also keep the connection alive. It seems like you dont have the latest version (where did you get the link from), maybe just private message pbx_support, indicate your OS and then we'll send you the link with the latest build.

 

actually I got a link from SNOM One when I registered and I downloaded that Link. My OS is windows 2008 server (64 bit). Can you please send the link to teh latest build.

 

Thanks

 

Sanjeev.

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I have PM'd the link already. Can you please verify?

 

Thanks you very much. Now i am able to place outbound calls and there is no timeout! Now for the last leg of my project. supervised transfer. I am having issues. After I transfer the call, the consulatation party cannot hear anything. It is just noise.. I will post the log in another thread.

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