I understand you can find it under this text. as you can see, the call just disapeare, i see now that the bye appears when i hang up the polycom phone.
I hope this information helps. U 172.16.0.12:5060 -> 172.16.1.10:5090 INVITE sip:[email protected]:5090;user=phone SIP/2.0. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK2867f2a43260B0C9. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>. CSeq: 1 INVITE. Call-ID: [email protected]. Contact: <sip:[email protected]>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049. Supported: 100rel,replaces. Allow-Events: talk,hold,conference. Max-Forwards: 70. Content-Type: application/sdp. Content-Length: 247. . v=0. o=- 1254823487 1254823487 IN IP4 172.16.0.12. s=Polycom IP Phone. c=IN IP4 172.16.0.12. t=0 0. m=audio 2222 RTP/AVP 0 8 18 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. U 172.16.1.10:5090 -> 172.16.0.12:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK2867f2a43260B0C9. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>;tag=67747a5e755302d1b99e6b9647717b58.aaf5. CSeq: 1 INVITE. Call-ID: [email protected]. Proxy-Authenticate: Digest realm="172.16.1.10", nonce="4acb4a6f000000003752ab36118b66f10da90b7ccd55e7e5". Server: OpenSIPS (1.5.3-notls (i386/linux)). Content-Length: 0. . U 172.16.0.12:5060 -> 172.16.1.10:5090 ACK sip:[email protected]:5090 SIP/2.0. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK2867f2a43260B0C9. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>;tag=67747a5e755302d1b99e6b9647717b58.aaf5. CSeq: 1 ACK. Call-ID: [email protected]. Contact: <sip:[email protected]>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049. Max-Forwards: 70. Content-Length: 0. . U 172.16.0.12:5060 -> 172.16.1.10:5090 INVITE sip:[email protected]:5090;user=phone SIP/2.0. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>. CSeq: 2 INVITE. Call-ID: [email protected]. Contact: <sip:[email protected]>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049. Supported: 100rel,replaces. Allow-Events: talk,hold,conference. Proxy-Authorization: Digest username="701", realm="172.16.1.10", nonce="4acb4a6f000000003752ab36118b66f10da90b7ccd55e7e5", uri="sip:[email protected]:5090;user=phone", response="7c019127f962f84b63ec55b27daac6e2", algorithm=MD5. Max-Forwards: 70. Content-Type: application/sdp. Content-Length: 247. . v=0. o=- 1254823487 1254823487 IN IP4 172.16.0.12. s=Polycom IP Phone. c=IN IP4 172.16.0.12. t=0 0. m=audio 2222 RTP/AVP 0 8 18 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. U 172.16.1.10:5090 -> 172.16.0.12:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>. CSeq: 2 INVITE. Call-ID: [email protected]. Server: OpenSIPS (1.5.3-notls (i386/linux)). Content-Length: 0. . U 172.16.1.10:5090 -> 172.16.0.12:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C. Record-Route: <sip:172.16.1.10:5090;lr=on>. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>;tag=as431f0138. Call-ID: [email protected]. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Contact: <sip:[email protected]>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 5484 5484 IN IP4 172.16.1.10. s=session. c=IN IP4 172.16.1.10. t=0 0. m=audio 17896 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=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 172.16.1.10:5090 -> 172.16.0.12:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C. Record-Route: <sip:172.16.1.10:5090;lr=on>. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>;tag=as431f0138. Call-ID: [email protected]. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Contact: <sip:[email protected]>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 5484 5485 IN IP4 172.16.1.10. s=session. c=IN IP4 172.16.1.10. t=0 0. m=audio 17896 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=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 172.16.0.12:5060 -> 172.16.1.10:5090 ACK sip:[email protected] SIP/2.0. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK1b9cbb48F2BE247D. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>;tag=as431f0138. Route: <sip:172.16.1.10:5090;lr=on>. CSeq: 2 ACK. Call-ID: [email protected]. Contact: <sip:[email protected]>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049. Proxy-Authorization: Digest username="701", realm="172.16.1.10", nonce="4acb4a6f000000003752ab36118b66f10da90b7ccd55e7e5", uri="sip:[email protected]:5090;user=phone", response="7c019127f962f84b63ec55b27daac6e2", algorithm=MD5. Max-Forwards: 70. Content-Length: 0. . U 172.16.1.10:5090 -> 172.16.0.12:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C. Record-Route: <sip:172.16.1.10:5090;lr=on>. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>;tag=as431f0138. Call-ID: [email protected]. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Contact: <sip:[email protected]>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 5484 5485 IN IP4 172.16.1.10. s=session. c=IN IP4 172.16.1.10. t=0 0. m=audio 17896 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=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 172.16.1.10:5090 -> 172.16.0.12:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C. Record-Route: <sip:172.16.1.10:5090;lr=on>. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>;tag=as431f0138. Call-ID: [email protected]. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Contact: <sip:[email protected]>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 5484 5485 IN IP4 172.16.1.10. s=session. c=IN IP4 172.16.1.10. t=0 0. m=audio 17896 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=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 172.16.1.10:5090 -> 172.16.0.12:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C. Record-Route: <sip:172.16.1.10:5090;lr=on>. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>;tag=as431f0138. Call-ID: [email protected]. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Contact: <sip:[email protected]>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 5484 5485 IN IP4 172.16.1.10. s=session. c=IN IP4 172.16.1.10. t=0 0. m=audio 17896 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=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 172.16.0.12:5060 -> 172.16.1.10:5090 BYE sip:[email protected] SIP/2.0. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bKcd927226B5124893. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>;tag=as431f0138. Route: <sip:172.16.1.10:5090;lr=on>. CSeq: 3 BYE. Call-ID: [email protected]. Contact: <sip:[email protected]>. User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049. Proxy-Authorization: Digest username="701", realm="172.16.1.10", nonce="4acb4a6f000000003752ab36118b66f10da90b7ccd55e7e5", uri="sip:[email protected]:5090;user=phone", response="9f5e7c543f689494d444f0402a1eca13", algorithm=MD5. Max-Forwards: 70. Content-Length: 0. . U 172.16.1.10:5090 -> 172.16.0.12:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bKcd927226B5124893. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>;tag=as431f0138. CSeq: 3 BYE. Call-ID: [email protected]. Server: OpenSIPS (1.5.3-notls (i386/linux)). Content-Length: 0. . 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=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 172.16.0.12:5060 -> 172.16.1.10:5090 ACK sip:[email protected] SIP/2.0. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK1b9cbb48F2BE247D. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>;tag=as431f0138. Route: <sip:172.16.1.10:5090;lr=on>. CSeq: 2 ACK. Call-ID: [email protected]. Contact: <sip:[email protected]>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049. Proxy-Authorization: Digest username="701", realm="172.16.1.10", nonce="4acb4a6f000000003752ab36118b66f10da90b7ccd55e7e5", uri="sip:[email protected]:5090;user=phone", response="7c019127f962f84b63ec55b27daac6e2", algorithm=MD5. Max-Forwards: 70. Content-Length: 0. . U 172.16.1.10:5090 -> 172.16.0.12:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C. Record-Route: <sip:172.16.1.10:5090;lr=on>. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>;tag=as431f0138. Call-ID: [email protected]. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Contact: <sip:[email protected]>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 5484 5485 IN IP4 172.16.1.10. s=session. c=IN IP4 172.16.1.10. t=0 0. m=audio 17896 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=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 172.16.1.10:5090 -> 172.16.0.12:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C. Record-Route: <sip:172.16.1.10:5090;lr=on>. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>;tag=as431f0138. Call-ID: [email protected]. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Contact: <sip:[email protected]>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 5484 5485 IN IP4 172.16.1.10. s=session. c=IN IP4 172.16.1.10. t=0 0. m=audio 17896 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=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 172.16.1.10:5090 -> 172.16.0.12:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C. Record-Route: <sip:172.16.1.10:5090;lr=on>. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>;tag=as431f0138. Call-ID: [email protected]. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Contact: <sip:[email protected]>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 5484 5485 IN IP4 172.16.1.10. s=session. c=IN IP4 172.16.1.10. t=0 0. m=audio 17896 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=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 172.16.0.12:5060 -> 172.16.1.10:5090 BYE sip:[email protected] SIP/2.0. Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bKcd927226B5124893. From: "701" <sip:[email protected]>;tag=A2EA31C5-DC95458E. To: <sip:[email protected];user=phone>;tag=as431f0138. Route: <sip:172.16.1.10:5090;lr=on>. CSeq: 3 BYE. Call-ID: [email protected]. Contact: <sip:[email protected]>. User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049. Proxy-Authorization: Digest username="701", realm="172.16.1.10", nonce="4acb4a6f00000000 Brett Nemeroff wrote: > > A trace of the whole call setup to hangup would be very helpful > > On Tue, Oct 6, 2009 at 8:31 AM, Peter den Hartog > <[email protected]>wrote: > >> >> Thanks alot for you reply. >> >> Asterisk is used because we have some agi stuff happening on incomming >> calls. The sip trunk is registered on Asterisk. If i dial out, opensips >> uses >> Asterisk because the extention is not in opensips (if i understand it >> correctly) then Asterisk just uses his own sip trunk to dial outside. >> >> But for me it would be fine to use Opensips directly to make the >> connection >> with the sip trunk, we can leave asterisk out for now. >> >> 1. There is two way audio, i can hear the other person talking, and he >> can >> hear me 2. >> 2. no reinvite, i see a ok, and then a bye >> 3. i don't know this yet, i can test it, i think i saw a empty ACK >> >> >> >> Brett Nemeroff wrote: >> > >> > I guess the question here is, what is asterisk doing for you? I >> personally >> > would prefer the sip trunks right on opensips.. Asterisk is a kinda >> funny >> > bottleneck in your architecture unless it's acting as some sort of >> media >> > server (or TDM gateway). >> > Some potential issues: >> > 1. Do you have 2 way audio, some providers (gateways) will disco the >> call >> > if >> > there is one way audio for X seconds. >> > 2. Do you see any reinvites happening? Some providers will re-invite >> calls >> > after they are up and if the reinvite fails, it will tear down the >> call. >> > 3. Where is the BYE coming from? Do you see any other signaling after >> the >> > 200OK/ACKs you get? Do you see retransmissions of either the 200OK or >> ACK? >> > If the signaling indicating the call was connected doesn't finish a >> proper >> > ACK in both directions, the call will likely get hung up on. >> > >> > >> > On Tue, Oct 6, 2009 at 8:17 AM, Peter den Hartog >> > <[email protected]>wrote: >> > >> >> >> >> I'm trying to intergrate opensips with a allready running Asterisk >> >> server. >> >> The two servers are both on the same machine. >> >> >> >> I can recieve calls fine, Asterisk send them to my opensips >> installation, >> >> and the opensips forwards the phone call to the right user. I can call >> >> between the users on the network, with out any issue's so far so good. >> >> >> >> I have a sip trunk registered on Asterisk, and i use that for my in >> and >> >> outgoing calls. >> >> >> >> But when i make an outside call, the call ends after 17 seconds. >> Looking >> >> at >> >> the sip messages i see that i recieve a bye, then the call is gone. >> >> >> >> Am i doing something wrong, should the sip trunk be directly in >> opensips? >> >> and add that as a rewritehost? Or is this an Asterisk issue? >> >> >> >> My opensips is running on port 5090 (so are the phones) and my >> >> asterisk+outside trunk is on 5060. >> >> -- >> >> View this message in context: >> >> >> http://n2.nabble.com/17-sec-recieve-a-bye-and-a-hangup-tp3774964p3774964.html >> >> Sent from the OpenSIPS - Users mailing list archive at Nabble.com. >> >> >> >> _______________________________________________ >> >> Users mailing list >> >> [email protected] >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > >> > _______________________________________________ >> > Users mailing list >> > [email protected] >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > >> > >> >> -- >> View this message in context: >> http://n2.nabble.com/17-sec-recieve-a-bye-and-a-hangup-tp3774964p3775048.html >> Sent from the OpenSIPS - Users mailing list archive at Nabble.com. >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- View this message in context: http://n2.nabble.com/17-sec-recieve-a-bye-and-a-hangup-tp3774964p3775118.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
