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Calls redirected to Speech Server do not hang up


satish_infy
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Pretty simple topology for demo purposes.. basically I have PBXnSIP 3.0 and Speech Server 07 on the same box.

 

The intent is to use PBXnSIP as a SIP gatway to speech server, from my softphones like Xlite or SJPhone.

 

I have setup two trunks, one for accepting the incoming SIP connection from my softphone, and another trunk to Speech Server (basically following these instructions http://www.selectec.co.uk/pbx.mht)

 

I have setup a dial-plan, and the SpeechServer trunk is configured for an outbound proxy.

 

Everything finally works.. only one problem, that PBXnSIP doesn't hangup the call, even though Speech Server already has.

 

Notice SpeechServer sent the BYE at 00:00:15

[7] 2008/08/24 00:00:15: SIP Rx tcp:10.156.112.97:19098:

BYE sip:satish@127.0.0.1:19106;transport=tcp SIP/2.0

FROM: <sip:333@localhost;user=phone>;epid=E9C4EF8C22;tag=6c6ad6a0b7

TO: <sip:satish@192.168.1.101;user=phone>;tag=41349

CSEQ: 1 BYE

CALL-ID: d40833a0@pbx

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 10.156.112.97:19098;branch=z9hG4bK4a1f364

CONTENT-LENGTH: 0

USER-AGENT: RTCC/3.0.0.0

 

And I had to manually hangup the client at 00:00:42

[7] 2008/08/24 00:00:42: SIP Rx tcp:10.110.8.81:4620:

BYE sip:333@10.156.112.97:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 10.110.8.81:48800;branch=z9hG4bK-d8754z-fe488c314e532a3d-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:satish@10.110.8.81:48800;transport=TCP>

To: "welcome To Ocs"<sip:333@10.156.112.97;transport=tcp>;tag=a8536b8484

From: "satish"<sip:satish@192.168.1.101>;tag=714c423f

Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.

CSeq: 2 BYE

User-Agent: X-Lite release 1100l stamp 47546

Reason: SIP;description="User Hung Up"

Content-Length: 0

 

Looks like the client never got the BYE at 00:15..

 

Attached is the log:

 
[5] 2008/08/24 00:00:08: SIP port accept from 10.110.8.81:4620 
[7] 2008/08/24 00:00:08: SIP Rx tcp:10.110.8.81:4620: 
INVITE sip:333@10.156.112.97;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.110.8.81:48800;branch=z9hG4bK-d8754z-f158cf53ab637e16-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:satish@10.110.8.81:48800;transport=TCP>
To: "welcome To Ocs"<sip:333@10.156.112.97;transport=tcp>
From: "satish"<sip:satish@192.168.1.101>;tag=714c423f
Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 471

v=0
o=- 0 2 IN IP4 192.168.1.101
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.101
t=0 0
m=audio 7150 RTP/AVP 107 119 100 106 0 105 98 8 3 101
a=alt:1 2 : twqBdPxb 9cMwUw1K 10.110.8.81 7150
a=alt:2 1 : azi2oQJs n+GjKaE4 192.168.1.101 7150
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

[7] 2008/08/24 00:00:08: UDP: Opening socket on port 61328 
[7] 2008/08/24 00:00:08: UDP: Opening socket on port 61329 
[5] 2008/08/24 00:00:08: Identify trunk (domain name match) 3 
[9] 2008/08/24 00:00:08: Resolve 167: tcp 10.110.8.81 4620 
[7] 2008/08/24 00:00:08: SIP Tx tcp:10.110.8.81:4620: 
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.110.8.81:48800;branch=z9hG4bK-d8754z-f158cf53ab637e16-1---d8754z-;rport=4620
From: "satish" <sip:satish@192.168.1.101>;tag=714c423f
To: "welcome To Ocs" <sip:333@10.156.112.97;transport=tcp>;tag=a8536b8484
Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.
CSeq: 1 INVITE
Content-Length: 0


[6] 2008/08/24 00:00:08: Sending RTP for Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.#a8536b8484 to 192.168.1.101:7150 
[8] 2008/08/24 00:00:08: Trunk: Changing the user to 999 
[9] 2008/08/24 00:00:08: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 333@10.156.112.97 
[5] 2008/08/24 00:00:08: Dialplan mss07: Match 333@10.156.112.97 to <sip:333@localhost;user=phone> on trunk SpeechServer 
[5] 2008/08/24 00:00:08: Using "satish" <sip:satish@192.168.1.101;user=phone>;tag=714c423f as redirect from 
[5] 2008/08/24 00:00:08: Charge user 999 for redirecting calls 
[8] 2008/08/24 00:00:08: Play audio_moh/noise.wav 
[7] 2008/08/24 00:00:08: UDP: Opening socket on port 49896 
[7] 2008/08/24 00:00:08: UDP: Opening socket on port 49897 
[9] 2008/08/24 00:00:08: Resolve 168: url sip:10.156.112.97:6060;transport=tcp 
[9] 2008/08/24 00:00:08: Resolve 168: a tcp 10.156.112.97 6060 
[9] 2008/08/24 00:00:08: Resolve 168: tcp 10.156.112.97 6060 
[7] 2008/08/24 00:00:08: SIP Tx tcp:10.156.112.97:6060: 
INVITE sip:333@localhost;user=phone SIP/2.0
Via: SIP/2.0/TCP 127.0.0.1:19105;branch=z9hG4bK-1476699eba8840bb676b659edb916c27;rport
From: "satish" <sip:satish@192.168.1.101;user=phone>;tag=41349
To: <sip:333@localhost;user=phone>
Call-ID: d40833a0@pbx
CSeq: 22698 INVITE
Max-Forwards: 70
Contact: <sip:satish@127.0.0.1:19105;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:999>;reason=unconditional;screen=no;privacy=off
P-Asserted-Identity: <sip:999@localhost;user=phone>
Content-Type: application/sdp
Content-Length: 284

v=0
o=- 52411 52411 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 49896 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=sendrecv

[9] 2008/08/24 00:00:08: Resolve 169: tcp 10.110.8.81 4620 
[7] 2008/08/24 00:00:08: SIP Tx tcp:10.110.8.81:4620: 
SIP/2.0 183 Ringing
Via: SIP/2.0/TCP 10.110.8.81:48800;branch=z9hG4bK-d8754z-f158cf53ab637e16-1---d8754z-;rport=4620
From: "satish" <sip:satish@192.168.1.101>;tag=714c423f
To: "welcome To Ocs" <sip:333@10.156.112.97;transport=tcp>;tag=a8536b8484
Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.
CSeq: 1 INVITE
Contact: <sip:333@10.156.112.97:5060;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
Content-Type: application/sdp
Content-Length: 241

v=0
o=- 56222 56222 IN IP4 10.156.112.97
s=-
c=IN IP4 10.156.112.97
t=0 0
m=audio 61328 RTP/AVP 0 8 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

[7] 2008/08/24 00:00:08: SIP Rx tcp:10.156.112.97:6060: 
SIP/2.0 100 Trying
FROM: "satish"<sip:satish@192.168.1.101;user=phone>;tag=41349
TO: <sip:333@localhost;user=phone>
CSEQ: 22698 INVITE
CALL-ID: d40833a0@pbx
VIA: SIP/2.0/TCP 127.0.0.1:19105;branch=z9hG4bK-1476699eba8840bb676b659edb916c27;rport
CONTENT-LENGTH: 0


[7] 2008/08/24 00:00:08: SIP Rx tcp:10.156.112.97:6060: 
SIP/2.0 302 Moved Temporarily
FROM: "satish"<sip:satish@192.168.1.101;user=phone>;tag=41349
TO: <sip:333@localhost;user=phone>;tag=a8d6c6f98
CSEQ: 22698 INVITE
CALL-ID: d40833a0@pbx
VIA: SIP/2.0/TCP 127.0.0.1:19105;branch=z9hG4bK-1476699eba8840bb676b659edb916c27;rport
CONTACT: <sip:333@localhost:19098;user=phone;transport=Tcp;maddr=10.156.112.97;x-mss-call-id=d40833a0%40pbx>
CONTENT-LENGTH: 0
SERVER: RTCC/3.0.0.0


[7] 2008/08/24 00:00:08: Call d40833a0@pbx#41349: Clear last INVITE 
[9] 2008/08/24 00:00:08: Resolve 170: url sip:10.156.112.97:6060;transport=tcp 
[9] 2008/08/24 00:00:08: Resolve 170: a tcp 10.156.112.97 6060 
[9] 2008/08/24 00:00:08: Resolve 170: tcp 10.156.112.97 6060 
[7] 2008/08/24 00:00:08: SIP Tx tcp:10.156.112.97:6060: 
ACK sip:333@localhost;user=phone SIP/2.0
Via: SIP/2.0/TCP 127.0.0.1:19105;branch=z9hG4bK-1476699eba8840bb676b659edb916c27;rport
From: "satish" <sip:satish@192.168.1.101;user=phone>;tag=41349
To: <sip:333@localhost;user=phone>;tag=a8d6c6f98
Call-ID: d40833a0@pbx
CSeq: 22698 ACK
Max-Forwards: 70
Contact: <sip:satish@127.0.0.1:19105;transport=tcp>
P-Asserted-Identity: <sip:999@localhost;user=phone>
Content-Length: 0


[5] 2008/08/24 00:00:08: Redirecting call 
[9] 2008/08/24 00:00:08: Resolve 171: aaaa tcp 10.156.112.97 19098 
[9] 2008/08/24 00:00:08: Resolve 171: a tcp 10.156.112.97 19098 
[9] 2008/08/24 00:00:08: Resolve 171: tcp 10.156.112.97 19098 
[7] 2008/08/24 00:00:08: SIP Tx tcp:10.156.112.97:19098: 
INVITE sip:333@localhost:19098;user=phone;transport=Tcp;maddr=10.156.112.97;x-mss-call-id=d40833a0%40pbx SIP/2.0
Via: SIP/2.0/TCP 127.0.0.1:19106;branch=z9hG4bK-a96f8438f6b5466df6a6c8968f199eb4;rport
From: "satish" <sip:satish@192.168.1.101;user=phone>;tag=41349
To: <sip:333@localhost;user=phone>
Call-ID: d40833a0@pbx
CSeq: 22699 INVITE
Max-Forwards: 70
Contact: <sip:satish@127.0.0.1:19106;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:999>;reason=unconditional;screen=no;privacy=off
P-Asserted-Identity: <sip:999@localhost;user=phone>
Content-Type: application/sdp
Content-Length: 284

v=0
o=- 52411 52411 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 49896 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=sendrecv

[7] 2008/08/24 00:00:09: SIP Rx tcp:10.156.112.97:19098: 
SIP/2.0 100 Trying
FROM: "satish"<sip:satish@192.168.1.101;user=phone>;tag=41349
TO: <sip:333@localhost;user=phone>
CSEQ: 22699 INVITE
CALL-ID: d40833a0@pbx
VIA: SIP/2.0/TCP 127.0.0.1:19106;branch=z9hG4bK-a96f8438f6b5466df6a6c8968f199eb4;rport
CONTENT-LENGTH: 0


[6] 2008/08/24 00:00:09: Sending RTP for Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.#a8536b8484 to 10.110.8.81:7150 
[7] 2008/08/24 00:00:09: SIP Rx tcp:10.156.112.97:19098: 
SIP/2.0 180 Ringing
FROM: "satish"<sip:satish@192.168.1.101;user=phone>;tag=41349
TO: <sip:333@localhost;user=phone>;epid=E9C4EF8C22;tag=6c6ad6a0b7
CSEQ: 22699 INVITE
CALL-ID: d40833a0@pbx
VIA: SIP/2.0/TCP 127.0.0.1:19106;branch=z9hG4bK-a96f8438f6b5466df6a6c8968f199eb4;rport
CONTENT-LENGTH: 0
SERVER: RTCC/3.0.0.0


[8] 2008/08/24 00:00:09: Play audio_en/ringback.wav 
[6] 2008/08/24 00:00:09: Sending RTP for d40833a0@pbx#41349 to 10.156.112.97:41728 
[7] 2008/08/24 00:00:10: SIP Rx tcp:10.156.112.97:19098: 
SIP/2.0 200 OK
FROM: "satish"<sip:satish@192.168.1.101;user=phone>;tag=41349
TO: <sip:333@localhost;user=phone>;epid=E9C4EF8C22;tag=6c6ad6a0b7
CSEQ: 22699 INVITE
CALL-ID: d40833a0@pbx
VIA: SIP/2.0/TCP 127.0.0.1:19106;branch=z9hG4bK-a96f8438f6b5466df6a6c8968f199eb4;rport
CONTACT: <sip:retlarch.ad.infosys.com:19098;transport=Tcp;maddr=10.156.112.97>;automata
CONTENT-LENGTH: 196
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 10.156.112.97
s=Microsoft Speech Server session
c=IN IP4 10.156.112.97
t=0 0
m=audio 41728 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

[7] 2008/08/24 00:00:10: Call d40833a0@pbx#41349: Clear last INVITE 
[7] 2008/08/24 00:00:10: Set packet length to 20 
[9] 2008/08/24 00:00:10: Resolve 172: aaaa tcp 10.156.112.97 19098 
[9] 2008/08/24 00:00:10: Resolve 172: a tcp 10.156.112.97 19098 
[9] 2008/08/24 00:00:10: Resolve 172: tcp 10.156.112.97 19098 
[7] 2008/08/24 00:00:10: SIP Tx tcp:10.156.112.97:19098: 
ACK sip:retlarch.ad.infosys.com:19098;transport=Tcp;maddr=10.156.112.97 SIP/2.0
Via: SIP/2.0/TCP 127.0.0.1:19106;branch=z9hG4bK-e8db414c4859d96f2949ac0c16fccf9c;rport
From: "satish" <sip:satish@192.168.1.101;user=phone>;tag=41349
To: <sip:333@localhost;user=phone>;tag=6c6ad6a0b7
Call-ID: d40833a0@pbx
CSeq: 22699 ACK
Max-Forwards: 70
Contact: <sip:satish@127.0.0.1:19106;transport=tcp>
P-Asserted-Identity: <sip:999@localhost;user=phone>
Content-Length: 0


[7] 2008/08/24 00:00:10: Determine pass-through mode after receiving response 
[9] 2008/08/24 00:00:10: Resolve 173: tcp 10.110.8.81 4620 
[7] 2008/08/24 00:00:10: SIP Tx tcp:10.110.8.81:4620: 
SIP/2.0 200 Ok
Via: SIP/2.0/TCP 10.110.8.81:48800;branch=z9hG4bK-d8754z-f158cf53ab637e16-1---d8754z-;rport=4620
From: "satish" <sip:satish@192.168.1.101>;tag=714c423f
To: "welcome To Ocs" <sip:333@10.156.112.97;transport=tcp>;tag=a8536b8484
Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.
CSeq: 1 INVITE
Contact: <sip:333@10.156.112.97:5060;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
Content-Type: application/sdp
Content-Length: 241

v=0
o=- 56222 56222 IN IP4 10.156.112.97
s=-
c=IN IP4 10.156.112.97
t=0 0
m=audio 61328 RTP/AVP 0 8 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

[7] 2008/08/24 00:00:10: d40833a0@pbx#41349: RTP pass-through mode 
[7] 2008/08/24 00:00:10: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.#a8536b8484: RTP pass-through mode 
[7] 2008/08/24 00:00:10: SIP Rx tcp:10.110.8.81:4620: 
ACK sip:333@10.156.112.97:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.110.8.81:48800;branch=z9hG4bK-d8754z-5d715935b7300d60-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:satish@10.110.8.81:48800;transport=TCP>
To: "welcome To Ocs"<sip:333@10.156.112.97;transport=tcp>;tag=a8536b8484
From: "satish"<sip:satish@192.168.1.101>;tag=714c423f
Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.
CSeq: 1 ACK
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 0


[7] 2008/08/24 00:00:15: SIP Rx tcp:10.156.112.97:19098: 
BYE sip:satish@127.0.0.1:19106;transport=tcp SIP/2.0
FROM: <sip:333@localhost;user=phone>;epid=E9C4EF8C22;tag=6c6ad6a0b7
TO: <sip:satish@192.168.1.101;user=phone>;tag=41349
CSEQ: 1 BYE
CALL-ID: d40833a0@pbx
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 10.156.112.97:19098;branch=z9hG4bK4a1f364
CONTENT-LENGTH: 0
USER-AGENT: RTCC/3.0.0.0


[9] 2008/08/24 00:00:15: Resolve 174: tcp 10.156.112.97 19098 
[7] 2008/08/24 00:00:15: SIP Tx tcp:10.156.112.97:19098: 
SIP/2.0 200 Ok
Via: SIP/2.0/TCP 10.156.112.97:19098;branch=z9hG4bK4a1f364
From: <sip:333@localhost;user=phone>;tag=6c6ad6a0b7;epid=E9C4EF8C22
To: <sip:satish@192.168.1.101;user=phone>;tag=41349
Call-ID: d40833a0@pbx
CSeq: 1 BYE
Contact: <sip:satish@127.0.0.1:19106;transport=tcp>
User-Agent: pbxnsip-PBX/3.0.0.2998
RTP-RxStat: Dur=7,Pkt=277,Oct=47644,Underun=0
RTP-TxStat: Dur=6,Pkt=285,Oct=49020
Content-Length: 0


[7] 2008/08/24 00:00:15: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.#a8536b8484: Media-aware pass-through mode 
[7] 2008/08/24 00:00:15: Other Ports: 1 
[7] 2008/08/24 00:00:15: Call Port: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.#a8536b8484 
[9] 2008/08/24 00:00:15: Resolve 175: url sip:satish@10.110.8.81:48800;transport=TCP 
[9] 2008/08/24 00:00:15: Resolve 175: a tcp 10.110.8.81 48800 
[9] 2008/08/24 00:00:15: Resolve 175: tcp 10.110.8.81 48800 
[7] 2008/08/24 00:00:15: SIP Tx tcp:10.110.8.81:48800: 
BYE sip:satish@10.110.8.81:48800;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 10.156.112.97:19109;branch=z9hG4bK-ee49be524ff19fe16e8ab0df1fd0a7a3;rport
From: "welcome To Ocs" <sip:333@10.156.112.97;transport=tcp>;tag=a8536b8484
To: "satish" <sip:satish@192.168.1.101>;tag=714c423f
Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.
CSeq: 23906 BYE
Max-Forwards: 70
Contact: <sip:333@10.156.112.97:19109;transport=tcp>
RTP-RxStat: Dur=7,Pkt=331,Oct=56932,Underun=0
RTP-TxStat: Dur=6,Pkt=328,Oct=56416
Content-Length: 0


[7] 2008/08/24 00:00:42: SIP Rx tcp:10.110.8.81:4620: 
BYE sip:333@10.156.112.97:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.110.8.81:48800;branch=z9hG4bK-d8754z-fe488c314e532a3d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:satish@10.110.8.81:48800;transport=TCP>
To: "welcome To Ocs"<sip:333@10.156.112.97;transport=tcp>;tag=a8536b8484
From: "satish"<sip:satish@192.168.1.101>;tag=714c423f
Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.
CSeq: 2 BYE
User-Agent: X-Lite release 1100l stamp 47546
Reason: SIP;description="User Hung Up"
Content-Length: 0


[9] 2008/08/24 00:00:42: Resolve 176: tcp 10.110.8.81 4620 
[7] 2008/08/24 00:00:42: SIP Tx tcp:10.110.8.81:4620: 
SIP/2.0 200 Ok
Via: SIP/2.0/TCP 10.110.8.81:48800;branch=z9hG4bK-d8754z-fe488c314e532a3d-1---d8754z-;rport=4620
From: "satish" <sip:satish@192.168.1.101>;tag=714c423f
To: "welcome To Ocs" <sip:333@10.156.112.97;transport=tcp>;tag=a8536b8484
Call-ID: Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.
CSeq: 2 BYE
Contact: <sip:333@10.156.112.97:5060;transport=tcp>
User-Agent: pbxnsip-PBX/3.0.0.2998
RTP-RxStat: Dur=34,Pkt=1632,Oct=280704,Underun=0
RTP-TxStat: Dur=32,Pkt=1653,Oct=284316
Content-Length: 0


[8] 2008/08/24 00:00:42: Hangup: Call Mjg0NDU0YTA3YWQ3YjEwMTI5NTg1YjE3OGRiMjA1ZjI.#a8536b8484 not found 

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Pretty simple topology for demo purposes.. basically I have PBXnSIP 3.0 and Speech Server 07 on the same box.

 

The intent is to use PBXnSIP as a SIP gatway to speech server, from my softphones like Xlite or SJPhone.

 

I have setup two trunks, one for accepting the incoming SIP connection from my softphone, and another trunk to Speech Server (basically following these instructions http://www.selectec.co.uk/pbx.mht)

 

I have setup a dial-plan, and the SpeechServer trunk is configured for an outbound proxy.

 

Everything finally works.. only one problem, that PBXnSIP doesn't hangup the call, even though Speech Server already has.

 

The problem is not on the speech server side. The PBX does try to send a BYE to the phone, but there is nothing coming back. See the BYE packet that the PBX tries to send right after receiving the BYE from the PBX.

 

I think it must have to do with the TCP transport layer. Just for the sake of locating the problem, can you chang ethe X-lite to UDP transport layer and see if the behavior changes?

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Awesome support PBXnSIP.. i did not expect such a quick reply.. you just won yourselves many client recommendations :angry:

 

You are right that PBXnSIP did transmit the BYE to the client, but the client never received it, and no ACK was seen in the pbxnsip log.

 

I changed xlite to use UDP, and bang, it hung up! thanks!!

 

But i'm technically curious.. why isn't tcp working.. all the other SIP commands before, like the INVITE, were in TCP right?

 

There is a VPN connection (with a firewall) in between.. my client VPN's to my corporate network where Speech Server is installed.. could it be that the VPN is blocking the BYE as it originated from the server? But shouldn't the firewall filter UDP traffic too then..?

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Awesome support PBXnSIP.. i did not expect such a quick reply.. you just won yourselves many client recommendations :angry:

 

:blush:

 

You are right that PBXnSIP did transmit the BYE to the client, but the client never received it, and no ACK was seen in the pbxnsip log.

 

I changed xlite to use UDP, and bang, it hung up! thanks!!

 

But i'm technically curious.. why isn't tcp working.. all the other SIP commands before, like the INVITE, were in TCP right?

 

There is a VPN connection (with a firewall) in between.. my client VPN's to my corporate network where Speech Server is installed.. could it be that the VPN is blocking the BYE as it originated from the server? But shouldn't the firewall filter UDP traffic too then..?

 

Well, you never really now. But I would give VPN only 1 % problem probability.

 

TCP for user agents was specified by the IETF in a sick way (they obviously cared more about the proxies). The idea was essentially that you open two TCP connections for each direction, which is a little bit tricky if you come from behind NAT. The new outbound solves that problem. But the X-lite was written way before outbound was proposed. Anyway, the discussion is kind of historical, so I think we should be pragmatic here and just use UDP.

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well its just not XLite.. another softphone like SJPhone using tcp had the same issue.. as soon as i used UDP (the default protocol in both softphones), the softphone hung up.

 

I was using TCP because Speech Server insists on SIP over TCP, and the PBX's trunk has its outbound proxy setup that way, ie to force transport=tcp when talking to Speech Server. Looks like PBXnSIP very nicely converts UDP to TCP and vice versa to facilitate communication between server and client.

 

Where exactly is the NAT in this topology btw?

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Hi PBXnSIP, where is the NAT in this topology?

 

No NAT here. Just wanted to make the general point with TCP that the contact header in SIP is useless for a user-agent ("Contact: <sip:lala@ip;transport=tcp>"), especially behind NAT. Instead of that, they should have taken something like a connection ID.

 

But without NAT is is almost the same problem, because also phones with a routable address typically do not accept incoming TCP connections, because of DoS and general programming pain in embedded environments.

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