April 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

codecs initialization flags in endpoint modules
by Lele Forzani 03 Apr '09

03 Apr '09

Hello, I've been experimenting with the use of mod_dahdi_codec and other ways to perform external transcoding for codecs, and came up with noticing that transcoding resources seemed to be used up twice what I expected. That is and 2x the number of call legs, ending up to two encoder and two decoder instances per leg. So, I looked at the code and noticed almost every endpoint module does something like this (excerpt from mod_sofia, sofia_glue.c:~1800): if (switch_core_codec_init(&tech_pvt->read_codec, tech_pvt->iananame, tech_pvt->rm_fmtp, tech_pvt->rm_rate, tech_pvt->codec_ms, 1, SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE | tech_pvt->profile->codec_flags, NULL, switch_core_session_get_pool(tech_pvt->session)) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Can't load codec?\n"); switch_goto_status(SWITCH_STATUS_FALSE, end); } if (switch_core_codec_init(&tech_pvt->write_codec, tech_pvt->iananame, tech_pvt->rm_fmtp, tech_pvt->rm_rate, tech_pvt->codec_ms, 1, SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE | tech_pvt->profile->codec_flags, NULL, switch_core_session_get_pool(tech_pvt->session)) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Can't load codec?\n"); switch_goto_status(SWITCH_STATUS_FALSE, end); } The flags being SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE seems to be causing the apparent 'double' allocation of transcoding resources, and I fail to understand the need for both, in both cases. Could someone please spend a minute to explain? thanks lele

2 1

Re: [Freeswitch-users] mod_shout delay in trunk
by freeswitch-users＠digitaldan.com 03 Apr '09

03 Apr '09

I have my burst rate set to something low, 4096 right now. I also wrote a flash/flex app that has the same size buffer which results in the audio being heard immediately when connecting. As far as the audio being real time, the audio stream is about 6 seconds behind which I'm guessing is the result of the size of the lame buffers in the mod_shout modules (i'm using g.711 ulaw), I was going to look into that next week. Anyone have any thoughts about where else the delay may be happening? I hoping to get this down to around 2 seconds. D- ----- Original Message ----- From: "Rupa Schomaker" <rupa(a)rupa.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, April 2, 2009 8:05:38 AM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk 2009/4/2 Anthony Minessale < anthony.minessale(a)gmail.com > Its the buffering and startup of the shout stream taking up the time, HINT put the shoutcast stream into a local stream with a .loc file and then play that in the conference. Ah, that is easy enough! Though I think with icecast doing the burst_on_connect thingie there should be enough data (pushed much faster than real time) to fill FS's buffers. But that would require mod_shout to cooperate with that strategy. ie: on connect, drain the socket as fast as it can filling it's own buffers. Once it's own buffers are full start streaming. I think right now it drains the socket only as fast as it needs to. Or maybe not. <shrug/> I'll go the local stream route for now.... -- -Rupa _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

1 0

Slow freeswitch shutdown
by Szymon Olko 03 Apr '09

03 Apr '09

In last SVN trunk version i noticed that stopping of freeswitch takes much time. I have configuration installed with freeswitch. I added sip gateway to my asterisk instance. I don't use asterisk currently and my gateway definition is like that: <gateway name="429956"> <param name="username" value="429956"/> <param name="password" value="429956"/> <param name="proxy" value="10.0.0.248:5059"/> <param name="register" value="false"/> </gateway> Starting freeswitch and shutting it down for console with '...' brings following logs. 2009-04-03 11:58:03 [NOTICE] sofia_reg.c:75 sofia_reg_kill_reg() UN-Registering example.com 2009-04-03 11:58:03 [NOTICE] sofia.c:895 sofia_profile_thread_run() Waiting for worker thread 2009-04-03 11:58:03 [NOTICE] sofia_glue.c:3173 sofia_glue_del_profile() deleted gateway example.com 2009-04-03 11:58:03 [NOTICE] sofia_reg.c:75 sofia_reg_kill_reg() UN-Registering 429956 2009-04-03 11:58:33 [NOTICE] sofia.c:895 sofia_profile_thread_run() Waiting for worker thread 2009-04-03 11:58:33 [NOTICE] sofia_glue.c:3173 sofia_glue_del_profile() deleted gateway 429956 Asterisk was not run at all so it should not register to it, why it hangs to unregister it?

2 1

Use rates from lcr in nibblebill module
by Yuriy Ivzhenko (WP) 03 Apr '09

03 Apr '09

Hi, I want to use module lcr to find a best route and his rate , then make a call and bill on that rate with nibblebill module. lcr return variable "lcr_auto_route" that contains "[lcr_rate=xxx]" variable for new channel. To use nibblebill i need to set "nibble_rate" = "lcr_rate". What is best method to do that? Is there a way to do that with standard tools, without use external scripts? Thanks, Yuriy

2 2

Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls
by Alfonso Pinto 03 Apr '09

03 Apr '09

Hi guys, I've using asterisk as PSTN gateway. When a call arrives from PSTN, I send the call to freeswitch and this route the call to a SIP gateway. When caller cancels the call (hangups before callee answers), I get this on asterisk CLI: chan_sip.c:13056 handle_response: Remote host can't match request CANCEL to call '271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1'. Giving up. I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 This is the sip call flow: u 2009/04/01 21:59:26.402934 2.2.2.2:5060 -> 1.1.1.1:5060 INVITE sip:666666666@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>. Contact: <sip:999999999@2.2.2.2>. Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Wed, 01 Apr 2009 21:03:12 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 265. . v=0. o=root 29347 29347 IN IP4 2.2.2.2. s=session. c=IN IP4 2.2.2.2. t=0 0. m=audio 13846 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2009/04/01 21:59:26.403717 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>. Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Content-Length: 0. . U 2009/04/01 21:59:26.414810 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>;tag=ceKFmNU84B90c. Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Proxy-Authenticate: Digest realm="1.1.1.1", nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", algorithm=MD5, qop="auth". Content-Length: 0. . U 2009/04/01 21:59:26.415395 2.2.2.2:5060 -> 1.1.1.1:5060 ACK sip:666666666@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>;tag=ceKFmNU84B90c. Contact: <sip:999999999@2.2.2.2>. Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. CSeq: 102 ACK. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . U 2009/04/01 21:59:26.415648 2.2.2.2:5060 -> 1.1.1.1:5060 INVITE sip:666666666@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>. Contact: <sip:999999999@2.2.2.2>. Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. CSeq: 103 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", algorithm=MD5, uri="sip:666666666@1.1.1.1", nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, cnonce="47efcad4", nc=00000001. Date: Wed, 01 Apr 2009 21:03:12 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 265. . v=0. o=root 29347 29348 IN IP4 2.2.2.2. s=session. c=IN IP4 2.2.2.2. t=0 0. m=audio 13846 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2009/04/01 21:59:26.416181 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>. Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. CSeq: 103 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Content-Length: 0. . U 2009/04/01 21:59:26.426298 1.1.1.1:5060 -> 3.3.3.3:5060 INVITE sip:666666666@3.3.3.3 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. Max-Forwards: 69. From: "999999999" <sip:559066555@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. To: <sip:666666666@3.3.3.3>. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. CSeq: 113193247 INVITE. Contact: <sip:gw+primus@1.1.1.1:5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 387. Remote-Party-ID: "999999999" <sip:999999999@3.3.3.3>;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1342987860622345384 4847837355494891206 IN IP4 1.1.1.1. s=FreeSWITCH. c=IN IP4 1.1.1.1. t=0 0. m=audio 31050 RTP/AVP 18 4 3 9 0 8 101 13. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:3 GSM/8000. a=rtpmap:9 G722/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20. U 2009/04/01 21:59:26.505833 3.3.3.3:5060 -> 1.1.1.1:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. From: "999999999" <sip:559066555@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. To: <sip:666666666@3.3.3.3>;tag=731C8E54-1862. Date: Fri, 05 Jan 2001 07:46:57 GMT. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 113193247 INVITE. Allow-Events: telephone-event. Content-Length: 0. . U 2009/04/01 21:59:40.281136 3.3.3.3:5060 -> 1.1.1.1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. From: "999999999" <sip:559066555@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. To: <sip:666666666@3.3.3.3>;tag=731C8E54-1862. Date: Fri, 05 Jan 2001 07:46:57 GMT. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 113193247 INVITE. Allow-Events: telephone-event. Contact: <sip:666666666@3.3.3.3:5060>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 300. . v=0. o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. s=SIP Call. c=IN IP4 3.3.3.3. t=0 0. m=audio 19398 RTP/AVP 18 13 101. c=IN IP4 3.3.3.3. a=rtpmap:18 G729/8000. a=rtpmap:13 CN/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:40. U 2009/04/01 21:59:40.325170 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. CSeq: 103 INVITE. Contact: <sip:mod_sofia@1.1.1.1:5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 292. . v=0. o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. s=FreeSWITCH. c=IN IP4 1.1.1.1. t=0 0. m=audio 20620 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. U 2009/04/01 21:59:44.342267 2.2.2.2:5060 -> 1.1.1.1:5060 CANCEL sip:666666666@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport. From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>. Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. CSeq: 103 CANCEL. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . U 2009/04/01 21:59:44.342568 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 481 Call/Transaction Does Not Exist. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport=5060. From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. CSeq: 103 CANCEL. Content-Length: 0. . U 2009/04/01 22:00:01.758748 3.3.3.3:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. From: "999999999" <sip:559066555@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. To: <sip:666666666@3.3.3.3>;tag=731C8E54-1862. Date: Fri, 05 Jan 2001 07:46:57 GMT. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 113193247 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:666666666@3.3.3.3:5060>. Content-Type: application/sdp. Content-Length: 300. . v=0. o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. s=SIP Call. c=IN IP4 3.3.3.3. t=0 0. m=audio 19398 RTP/AVP 18 13 101. c=IN IP4 3.3.3.3. a=rtpmap:18 G729/8000. a=rtpmap:13 CN/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:40. U 2009/04/01 22:00:01.759460 1.1.1.1:5060 -> 3.3.3.3:5060 ACK sip:666666666@3.3.3.3:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKttg8r61Ka85yj. Max-Forwards: 70. From: "999999999" <sip:559066555@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. To: <sip:666666666@3.3.3.3>;tag=731C8E54-1862. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. CSeq: 113193247 ACK. Contact: <sip:gw+primus@1.1.1.1:5060;transport=udp>. Content-Length: 0. . U 2009/04/01 22:00:01.779058 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. CSeq: 103 INVITE. Contact: <sip:mod_sofia@1.1.1.1:5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 292. . v=0. o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. s=FreeSWITCH. c=IN IP4 1.1.1.1. t=0 0. m=audio 20620 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. U 2009/04/01 22:00:01.780198 2.2.2.2:5060 -> 1.1.1.1:5060 ACK sip:mod_sofia@1.1.1.1:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3de2cb0a;rport. From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. Contact: <sip:999999999@2.2.2.2>. Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. CSeq: 103 ACK. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . U 2009/04/01 22:00:01.780214 2.2.2.2:5060 -> 1.1.1.1:5060 BYE sip:mod_sofia@1.1.1.1:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport. From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. CSeq: 104 BYE. User-Agent: Asterisk PBX. Max-Forwards: 70. Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", algorithm=MD5, uri="sip:mod_sofia@1.1.1.1:5060", nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", response="21ee4a61f1751494e2e96254dd007a4c", qop=auth, cnonce="6bc43301", nc=00000002. Content-Length: 0. . U 2009/04/01 22:00:01.780814 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport=5060. From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. CSeq: 104 BYE. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Content-Length: 0. . U 2009/04/01 22:00:01.802580 1.1.1.1:5060 -> 3.3.3.3:5060 BYE sip:666666666@3.3.3.3:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. Max-Forwards: 70. From: "999999999" <sip:559066555@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. To: <sip:666666666@3.3.3.3>;tag=731C8E54-1862. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. CSeq: 113193248 BYE. Contact: <sip:gw+primus@1.1.1.1:5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. . U 2009/04/01 22:00:01.873399 3.3.3.3:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. From: "999999999" <sip:559066555@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. To: <sip:666666666@3.3.3.3>;tag=731C8E54-1862. Date: Fri, 05 Jan 2001 07:47:32 GMT. Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. Server: Cisco-SIPGateway/IOS-12.x. Content-Length: 0. CSeq: 113193248 BYE. . Please, can somebody tell me what is happening?. Thanks in advance. Regards.

4 9

Re: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls [SOLVED]
by Alfonso Pinto 03 Apr '09

03 Apr '09

Hi, Updating asterisk to version 1.4.24 solved the problem. Thanks guys. Regards. 2009/4/2 Brian West <brian(a)freeswitch.org>: > Follow this > thread http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012646.ht… > /b > On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: > > Hi guys, > > I've using asterisk as PSTN gateway. When a call arrives from PSTN, I > send the call to freeswitch and this route the call to a SIP gateway. > > When caller cancels the call (hangups before callee answers), I get > this on asterisk CLI: > > chan_sip.c:13056 handle_response: Remote host can't match request > CANCEL to call '271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1'. Giving up. > > I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 > > This is the sip call flow: > > u 2009/04/01 21:59:26.402934 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666@1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. > To: <sip:666666666@1.1.1.1>. > Contact: <sip:999999999@2.2.2.2>. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. > CSeq: 102 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29347 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.403717 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. > To: <sip:666666666@1.1.1.1>. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.414810 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 407 Proxy Authentication Required. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. > From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. > To: <sip:666666666@1.1.1.1>;tag=ceKFmNU84B90c. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. > CSeq: 102 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Proxy-Authenticate: Digest realm="1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", algorithm=MD5, > qop="auth". > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415395 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:666666666@1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. > From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. > To: <sip:666666666@1.1.1.1>;tag=ceKFmNU84B90c. > Contact: <sip:999999999@2.2.2.2>. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. > CSeq: 102 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.415648 2.2.2.2:5060 -> 1.1.1.1:5060 > INVITE sip:666666666@1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. > From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. > To: <sip:666666666@1.1.1.1>. > Contact: <sip:999999999@2.2.2.2>. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. > CSeq: 103 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:666666666@1.1.1.1", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="cb57576192b001f79bd03ebb8bb57d0a", qop=auth, > cnonce="47efcad4", nc=00000001. > Date: Wed, 01 Apr 2009 21:03:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 265. > . > v=0. > o=root 29347 29348 IN IP4 2.2.2.2. > s=session. > c=IN IP4 2.2.2.2. > t=0 0. > m=audio 13846 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2009/04/01 21:59:26.416181 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. > To: <sip:666666666@1.1.1.1>. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. > CSeq: 103 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Content-Length: 0. > . > > > U 2009/04/01 21:59:26.426298 1.1.1.1:5060 -> 3.3.3.3:5060 > INVITE sip:666666666@3.3.3.3 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > Max-Forwards: 69. > From: "999999999" <sip:559066555@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: <sip:666666666@3.3.3.3>. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 INVITE. > Contact: <sip:gw+primus@1.1.1.1:5060;transport=udp>. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 387. > Remote-Party-ID: "999999999" <sip:999999999@3.3.3.3>;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1342987860622345384 4847837355494891206 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 31050 RTP/AVP 18 4 3 9 0 8 101 13. > a=rtpmap:18 G729/8000. > a=rtpmap:4 G723/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:9 G722/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=rtpmap:13 CN/8000. > a=ptime:20. > > > U 2009/04/01 21:59:26.505833 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" <sip:559066555@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: <sip:666666666@3.3.3.3>;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Content-Length: 0. > . > > > U 2009/04/01 21:59:40.281136 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" <sip:559066555@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: <sip:666666666@3.3.3.3>;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow-Events: telephone-event. > Contact: <sip:666666666@3.3.3.3:5060>. > Content-Disposition: session;handling=required. > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 21:59:40.325170 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. > To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. > CSeq: 103 INVITE. > Contact: <sip:mod_sofia@1.1.1.1:5060;transport=udp>. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 21:59:44.342267 2.2.2.2:5060 -> 1.1.1.1:5060 > CANCEL sip:666666666@1.1.1.1 SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport. > From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. > To: <sip:666666666@1.1.1.1>. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. > CSeq: 103 CANCEL. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 21:59:44.342568 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 481 Call/Transaction Does Not Exist. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3912e266;rport=5060. > From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. > To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. > CSeq: 103 CANCEL. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.758748 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. > From: "999999999" <sip:559066555@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: <sip:666666666@3.3.3.3>;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:46:57 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 113193247 INVITE. > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, > SUBSCRIBE, NOTIFY, INFO. > Allow-Events: telephone-event. > Contact: <sip:666666666@3.3.3.3:5060>. > Content-Type: application/sdp. > Content-Length: 300. > . > v=0. > o=CiscoSystemsSIP-GW-UserAgent 6896 7915 IN IP4 3.3.3.3. > s=SIP Call. > c=IN IP4 3.3.3.3. > t=0 0. > m=audio 19398 RTP/AVP 18 13 101. > c=IN IP4 3.3.3.3. > a=rtpmap:18 G729/8000. > a=rtpmap:13 CN/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:40. > > > U 2009/04/01 22:00:01.759460 1.1.1.1:5060 -> 3.3.3.3:5060 > ACK sip:666666666@3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKttg8r61Ka85yj. > Max-Forwards: 70. > From: "999999999" <sip:559066555@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: <sip:666666666@3.3.3.3>;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193247 ACK. > Contact: <sip:gw+primus@1.1.1.1:5060;transport=udp>. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.779058 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. > From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. > To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. > CSeq: 103 INVITE. > Contact: <sip:mod_sofia@1.1.1.1:5060;transport=udp>. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 292. > . > v=0. > o=FreeSWITCH 2480261112724725981 6822082830258713617 IN IP4 1.1.1.1. > s=FreeSWITCH. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 20620 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > > > U 2009/04/01 22:00:01.780198 2.2.2.2:5060 -> 1.1.1.1:5060 > ACK sip:mod_sofia@1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3de2cb0a;rport. > From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. > To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. > Contact: <sip:999999999@2.2.2.2>. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. > CSeq: 103 ACK. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780214 2.2.2.2:5060 -> 1.1.1.1:5060 > BYE sip:mod_sofia@1.1.1.1:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport. > From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. > To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. > CSeq: 104 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Proxy-Authorization: Digest username="asterisk02", realm="1.1.1.1", > algorithm=MD5, uri="sip:mod_sofia@1.1.1.1:5060", > nonce="5df21692-1f08-11de-9d06-83e4a6e70df7", > response="21ee4a61f1751494e2e96254dd007a4c", qop=auth, > cnonce="6bc43301", nc=00000002. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.780814 1.1.1.1:5060 -> 2.2.2.2:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK3871369b;rport=5060. > From: "999999999" <sip:999999999@1.1.1.1>;tag=as26208773. > To: <sip:666666666@1.1.1.1>;tag=DQc8Ngcc2mZKr. > Call-ID: 271c0dad41cc80456b8de2133dc80b2e(a)1.1.1.1. > CSeq: 104 BYE. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.802580 1.1.1.1:5060 -> 3.3.3.3:5060 > BYE sip:666666666@3.3.3.3:5060 SIP/2.0. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > Max-Forwards: 70. > From: "999999999" <sip:559066555@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: <sip:666666666@3.3.3.3>;tag=731C8E54-1862. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > CSeq: 113193248 BYE. > Contact: <sip:gw+primus@1.1.1.1:5060;transport=udp>. > User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Reason: Q.850;cause=16;text="NORMAL_CLEARING". > Content-Length: 0. > . > > > U 2009/04/01 22:00:01.873399 3.3.3.3:5060 -> 1.1.1.1:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKU390t1jQ7gvHe. > From: "999999999" <sip:559066555@3.3.3.3;transport=udp>;tag=e050QBXFZXN6K. > To: <sip:666666666@3.3.3.3>;tag=731C8E54-1862. > Date: Fri, 05 Jan 2001 07:47:32 GMT. > Call-ID: 5df4fd12-1f08-11de-9d06-83e4a6e70df7. > Server: Cisco-SIPGateway/IOS-12.x. > Content-Length: 0. > CSeq: 113193248 BYE. > . > > Please, can somebody tell me what is happening?. > > Thanks in advance. > > Regards. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Brian West > brian(a)freeswitch.org > -- Meet us a ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

1 0

Handle invite with wrong to:IP
by Peter P GMX 02 Apr '09

02 Apr '09

Hello, I am using a SIP account from Netvoip CH. I try to receive inbound call from this SIP trunk. I discovered that, when they sent an invite, the IP-Adress of the to: is their own IP address. There fore ACL doesn't work and FS asks for authorization, which then fails I receive the following message on the CLI: 2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth() Can't find user [anonymous(a)62.65.128.62] You must define a domain called '62.65.128.62' in your directory and add a user with the id="anonymous" attribute and you must configure your device to use the proper domain in it's authentication credentials. I could do that, but this is not clean and I do not have a password for that. How can I workaround this, so that Freeswitch accepts this call? Aliases do not seem to work. Here is a sample message after FS asks for authorization: xx.xx.xxx.xxx is the IP of our Freeswitch 62.65.128.62 is the IP of Netvoip CH I would expect To: <sip:0715aaaaaa@xx.xx.xxx.xxx>. instead of To: <sip:0715aaaaaa@62.65.128.62>. U 62.65.128.62:5060 -> xx.xx.xxx.xxx:5080 INVITE sip:0715aaaaaa@xx.xx.xxx.xxx:5080 SIP/2.0. Via: SIP/2.0/UDP 62.65.128.62:5060. Via: SIP/2.0/UDP 62.65.128.61:5060;branch=z9hG4bK8c977d2613c4d7d1fd9d03d4. Max-Forwards: 69. From: <sip:0049xxxxxxxxxxx@62.65.128.62>;tag=8c977d2613672832fd9d03e9. To: <sip:0715aaaaaa@62.65.128.62>. Call-ID: 8c977d261329cd80fd9d03d6(a)62.65.128.61. CSeq: 2 INVITE. User-agent: Netstream VoIP Gateway. Contact: <sip:0049xxxxxxxxxxx@62.65.128.62:5060>. Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 584. Proxy-Authorization: Digest username="anonymous", realm="62.65.128.62", nonce="a4151ee0-1fbb-11de-b056-494b9de21e06", nc="00000001", uri="sip:0715aaaaaa@62.65.128.62:5060", cnonce="5f109eee", response="62faa6d38b3b12c3626753395a8b507c", algorithm="MD5", qop="auth". . v=0. o=- 225947743692042 1 IN IP4 62.65.128.62. s=-. c=IN IP4 62.65.128.62. t=0 0. m=audio 28224 RTP/AVP 8 18 4 3 100 100 99 100 100 98 97 96 105 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000. a=fmtp:4 annexa=no. a=rtpmap:3 GSM/8000. a=rtpmap:100 speex/8000. a=rtpmap:100 speex/8000. a=rtpmap:99 G726-16/8000. a=rtpmap:100 speex/8000. a=rtpmap:100 speex/8000. a=rtpmap:98 G726-24/8000. a=rtpmap:97 G726-32/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:105 iLBC/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. Best regards Peter

3 5

FreeSWITCH 1.0.4pre3 is slow to stop
by Ceino 02 Apr '09

02 Apr '09

Hi all, I have tested it a little bit and it's works well. But when I give it the command to stop (...) it use about 40 sec to stop (1.0.3 use about 5 sec). Here is a log over where is hang (looks like a Sofia thread use long time to stop): ------------------------------------------------------------------ 2009-04-02 21:43:41 [NOTICE] sofia.c:877 sofia_profile_thread_run() Waiting for worker thread 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock internal-ipv6 2009-04-02 21:43:41 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock internal 2009-04-02 21:44:11 [NOTICE] sofia.c:877 sofia_profile_thread_run() Waiting for worker thread 2009-04-02 21:44:11 [NOTICE] sofia_glue.c:3167 sofia_glue_del_profile() deleted gateway example.com 2009-04-02 21:44:11 [DEBUG] sofia.c:923 sofia_profile_thread_run() Write unlock external ------------------------------------------------------------------ Best Regards Lars Sivertsen

3 2

ANNOUNCEMENT: FreeSWITCH 1.0.4pre3 Now Available
by Michael Collins 02 Apr '09

02 Apr '09

The FreeSWITCH team would like to let everyone know that the latest version is available. More information can be found here: http://www.freeswitch.org/node/172 By all means download, upgrade, test, and report back! Your feedback helps us make FreeSWITCH even better! -MC

3 2

loopback-b channels stay alive
by Tamas Cseke 02 Apr '09

02 Apr '09

Hello, We originate loopback channels and they end up in calling sofia and transfer the call to a fifo. If we have a heavy call volume loopback-b channels don't hangup properly. They stay in core.db. Unfortunetly we can't reproduce it on test boxes but happens every day. On this box we had to turn off debug logging, becase we had I/O problems. The only thing I saw in log that switch_core_session_thread don't call switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "Session %" SWITCH_SIZE_T_FMT " (%s) Ended\n", session->id, switch_channel_get_name(session->channel)); in these cases. We have local patches (I don't think they are related) and we are running FS on virtual machine and we had some problem with that before so I'm not sure, but I guess it is maybe a lock or mutex problem. I tried SWITCH_DEBUG_RWLOCKS, but I got build error, and I don't know what to do with it. FS_CFLAGS = -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS export CFLAGS="-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS" export MOD_CFLAGS="-O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS" ./configure gcc -I/DEVEL/freeswitch/src/include -I/DEVEL/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -O2 -ffast-math -g -ggdb -DSWITCH_DEBUG_RWLOCKS -Wall -std=c99 -pedantic -o .libs/freeswitch freeswitch-switch.o -lm ./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a -lrt -ldl -lcrypt -lpthread libs/libedit/src/.libs/libedit.a -lncurses -Wl,--rpath -Wl,/opt/freeswitch//lib ./.libs/libfreeswitch.so: undefined reference to `switch_core_session_read_lock' ./.libs/libfreeswitch.so: undefined reference to `switch_core_session_locate' ./.libs/libfreeswitch.so: undefined reference to `switch_core_session_rwunlock' collect2: ld returned 1 exit status make[2]: *** [freeswitch] Error 1 Could you please tell me how could I test mutexes, rwlocks? Other option would be to omit loopback channels. Anthony earlier suggested me to avoid it and call sofia directly "you could make the loopback channel execute the eval app and do the originate to the sofia channel from the dialplan. <action application="eval" data="${originate(sofia/foo/a(a)b.com xyz)}"/> or make the loopback chan exec a lua or js and fire an originate command and exit This way you don't have the loopback a and b leg as well as the sofia chan." but it doesn't work, because originate api doesn't let us originate inside a session. So we still using it. Thanks in advance, Tamas

3 2