Summary: | ASTERISK-11471: Asterisk 1.6-beta3 does not follow sip redirect using sip/tcp | ||
Reporter: | Olaf Holthausen (oholthau) | Labels: | |
Date Opened: | 2008-02-19 05:52:21.000-0600 | Date Closed: | 2008-02-19 09:02:33.000-0600 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_sip/Interoperability |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | I tried to connect Asterisk 1.6 beta3 to a Microsoft Speech Server 2007. Microsoft Speech Server is by default using a sip redirect to another sip port to the same machine. Asterisk is always trying to use the default port (5060) instead of the new one. Microsoft SPeech Server is using the same IVR stack as Exchange 2007 Unified Messaging. I assume the Microsoft Solutions become quite popular in the future.. Best Regards Olaf. ****** ADDITIONAL INFORMATION ****** Here is the sip debug trace: -- Executing [00@asteriske1:1] Dial("SIP/10.25.31.142-08630fb0", "sip/20@10.25.31.141") in new stack == Using SIP RTP CoS mark 5 Audio is at 10.25.31.130 port 19064 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.25.31.141:5060: INVITE sip:20@10.25.31.141 SIP/2.0 Via: SIP/2.0/TCP 10.25.31.130:5060;branch=z9hG4bK66e879df;rport Max-Forwards: 70 From: "44" <sip:44@10.25.31.130>;tag=as0543e777 To: <sip:20@10.25.31.141> Contact: <sip:44@10.25.31.130:5060;transport=TCP> Call-ID: 1168b68c6a4ee209112d4fd40b153d05@10.25.31.130 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0-beta3 Date: Tue, 19 Feb 2008 11:43:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 432935516 432935516 IN IP4 10.25.31.130 s=Asterisk PBX 1.6.0-beta3 c=IN IP4 10.25.31.130 t=0 0 m=audio 19064 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 20@10.25.31.141 <--- SIP read from TCP://10.25.31.141:5060 ---> SIP/2.0 100 Trying FROM: "44"<sip:44@10.25.31.130>;tag=as0543e777 TO: <sip:20@10.25.31.141> CSEQ: 102 INVITE CALL-ID: 1168b68c6a4ee209112d4fd40b153d05@10.25.31.130 VIA: SIP/2.0/TCP 10.25.31.130:5060;branch=z9hG4bK66e879df;rport CONTENT-LENGTH: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from TCP://10.25.31.141:5060 ---> SIP/2.0 302 Moved Temporarily FROM: "44"<sip:44@10.25.31.130>;tag=as0543e777 TO: <sip:20@10.25.31.141>;tag=3ab9584f1 CSEQ: 102 INVITE CALL-ID: 1168b68c6a4ee209112d4fd40b153d05@10.25.31.130 VIA: SIP/2.0/TCP 10.25.31.130:5060;branch=z9hG4bK66e879df;rport CONTACT: <sip:20@10.25.31.141:1048;transport=Tcp;maddr=10.25.31.141;x-mss-call-id=1168b68c6a4ee209112d4fd40b153d05%4010.25.31.130> CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 <-------------> --- (9 headers 0 lines) --- -- Got SIP response 302 "Moved Temporarily" back from 10.25.31.141 -- Now forwarding SIP/10.25.31.142-08630fb0 to 'SIP/20@10.25.31.141:1048' (thanks to SIP/10.25.31.141-08628878) == Using SIP RTP CoS mark 5 Audio is at 10.25.31.130 port 13808 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.25.31.141:5060: INVITE sip:20@10.25.31.141 SIP/2.0 Via: SIP/2.0/TCP 10.25.31.130:5060;branch=z9hG4bK67449e2e;rport Max-Forwards: 70 From: "44" <sip:44@10.25.31.130>;tag=as7047f18e To: <sip:20@10.25.31.141> Contact: <sip:44@10.25.31.130:5060;transport=TCP> Call-ID: 03b618ed05ebd123567d5a735a375b84@10.25.31.130 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0-beta3 Date: Tue, 19 Feb 2008 11:43:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 87528796 87528796 IN IP4 10.25.31.130 s=Asterisk PBX 1.6.0-beta3 c=IN IP4 10.25.31.130 t=0 0 m=audio 13808 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Transmitting (no NAT) to 10.25.31.141:5060: ACK sip:20@10.25.31.141 SIP/2.0 Via: SIP/2.0/TCP 10.25.31.130:5060;branch=z9hG4bK66e879df;rport Max-Forwards: 70 From: "44" <sip:44@10.25.31.130>;tag=as0543e777 To: <sip:20@10.25.31.141>;tag=3ab9584f1 Contact: <sip:44@10.25.31.130:5060;transport=TCP> Call-ID: 1168b68c6a4ee209112d4fd40b153d05@10.25.31.130 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0-beta3 Content-Length: 0 --- <--- SIP read from TCP://10.25.31.141:5060 ---> SIP/2.0 100 Trying FROM: "44"<sip:44@10.25.31.130>;tag=as7047f18e TO: <sip:20@10.25.31.141> CSEQ: 102 INVITE CALL-ID: 03b618ed05ebd123567d5a735a375b84@10.25.31.130 VIA: SIP/2.0/TCP 10.25.31.130:5060;branch=z9hG4bK67449e2e;rport CONTENT-LENGTH: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from TCP://10.25.31.141:5060 ---> SIP/2.0 302 Moved Temporarily FROM: "44"<sip:44@10.25.31.130>;tag=as7047f18e TO: <sip:20@10.25.31.141>;tag=c436f6fff8 CSEQ: 102 INVITE CALL-ID: 03b618ed05ebd123567d5a735a375b84@10.25.31.130 VIA: SIP/2.0/TCP 10.25.31.130:5060;branch=z9hG4bK67449e2e;rport CONTACT: <sip:20@10.25.31.141:1048;transport=Tcp;maddr=10.25.31.141;x-mss-call-id=03b618ed05ebd123567d5a735a375b84%4010.25.31.130> CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 <-------------> --- (9 headers 0 lines) --- -- Got SIP response 302 "Moved Temporarily" back from 10.25.31.141 Transmitting (no NAT) to 10.25.31.141:5060: ACK sip:20@10.25.31.141 SIP/2.0 Via: SIP/2.0/TCP 10.25.31.130:5060;branch=z9hG4bK67449e2e;rport Max-Forwards: 70 From: "44" <sip:44@10.25.31.130>;tag=as7047f18e To: <sip:20@10.25.31.141>;tag=c436f6fff8 Contact: <sip:44@10.25.31.130:5060;transport=TCP> Call-ID: 03b618ed05ebd123567d5a735a375b84@10.25.31.130 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0-beta3 Content-Length: 0 --- Really destroying SIP dialog '1168b68c6a4ee209112d4fd40b153d05@10.25.31.130' Method: INVITE The sip.conf file: [general] tcpenable=yes ; Enable server for incoming TCP connections (default is yes) tcpbindaddr=0.0.0.0 ; IP adderss for TCP server to bind to (0.0.0.0 binds to all interfaces) promiscredir = yes ; If yes, allows 302 or REDIR to non-local SIP address [10.25.31.141] type=friend host=10.25.31.141 transport=tcp port=5060 disallow=all allow=alaw insecure=port,invite context=speechy | ||
Comments: | By: Joshua C. Colp (jcolp) 2008-02-19 09:02:33.000-0600 Closed as this is a duplicate of issue 11843. |