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FW: FS Does Not Relay PresencePUBLISHToSubscribing Phones
by Jerry Richards 09 Oct '09

09 Oct '09

I put the sqlite3 select query in the paste bin again, and prior to that, I entered the .dump command. The select command came back with the sqlite3 prompt, which I guess means it didn't find an entry. How do I go about isolating this problem? I'm using CounterPath's Bria Professional softphone. They are the same company that make the Eyebeam. Any ideas? Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Friday, October 02, 2009 11:28 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Does Not Relay PresencePUBLISHToSubscribing Phones I put the sqlite3 select query in the paste bin, and prior to that, I entered the .dump command. The select command came back with a "...>" prompt which I don't understand. I don't know enough about sqlite3 to know what that means? Best Regards, Jerry _____ From: Anthony Minessale [mailto:anthony.minessale@gmail.com] Sent: Friday, October 02, 2009 10:52 AM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay PresencePUBLISHToSubscribing Phones connect to sqlite directly with sqlite3 app and try that sql stmt and see why it doesn't match anything. sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name, 'Away','away','192.168.72.38',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where (event='presence' or event='presence') and sub_to_user='1001' and (sub_to_host='192.168.72.38' or presence_hosts like '%192.168.72.38%') and (sip_subscriptions.profile_name = 'external' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) On Fri, Oct 2, 2009 at 12:12 PM, Jerry Richards <jerry.richards(a)teotech.com> wrote: Okay, I put a log up on the pastebin that shows the PUBLISH event coming from a CounterPath Bria Professional phone. For some reason, FS is getting an error and not relaying the presence status to the subscriber. Best Regards, Jerry _____ From: João Mesquita [mailto:jmesquita@freeswitch.org] Sent: Thursday, October 01, 2009 8:14 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISHToSubscribing Phones Piece of advice, don't ask, just do it. ;) jmesquita On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards <jerry.richards(a)teotech.com> wrote: If you have time to take a look, I could put a trace in the pastebin? Jerry _____ From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Thursday, October 01, 2009 10:29 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones I am using two Bria Professional Version 2.5.4 Build 54835 softphones. Thanks, Jerry _____ From: Anthony Minessale [mailto:anthony.minessale@gmail.com] Sent: Thursday, October 01, 2009 9:36 AM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones which phone is it, we tested it with eyebeam and it appears to work for us. On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards <jerry.richards(a)teotech.com> wrote: By the way, I see the following lines at the FS console, which might be a clue as to why this is happening. Could someone point me toward what might cause this? I set the "manage-presence" parameter to "true" in each XML file where I saw it defined. [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [WARNING] sofia_presence.c:565 192.168.72.38 is an alias, skipping Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones I have two phones configured to subscribe to each other's presence status. When I change the presence status in one phone, I see the SIP PUBLISH message going to FS, but I don't see FS relaying that presence status to the subscribing phone. Does anyone know why? Best Regards, Jerry _______________________________________________ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ 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 _______________________________________________ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400

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FreeSWITCH Weekly Conference Call Starting, Please Join Us!
by Michael Collins 09 Oct '09

09 Oct '09

Agenda and call-in info: http://bit.ly/lzEYy

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NAT problems - sorry
by Andy 09 Oct '09

09 Oct '09

Hi folks, I know this problem comes up all the time so sorry to bring it up again but I can't seem to find the answer in previous posts. I have my freeswitch installation behind a DLink firewall so the freeswitch server is natted. I have added what I believe are all the necessary rules to the firewall to handle SIP correctly and the firewall isn't showing any blocked traffic when calls come in. Calls are forwarded to my public ip address from my VOIP provider Voiptalk. Here's the problem: Incoming calls are successfully forwarded to the freeswitch server and answered. I can hear the IVR message played down the phone line so outgoing audio is ok. When I try to send DTMF key presses from my phone none of these register on the switch. When I hangup the call the call hangup is also not picked up by the switch. I have followed the auto-nat instrcutions and have verified that the switch correctly grabs and sets the external ip address. I have also tried setting the external ip address manually but without success. Can anyone offer any guidance as to what might be causing this problem or what I could try to resolve it? I can post log/config snippets if that helps. Many thanks Andy

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[ERR] mod_dingaling.c:980 Stun Failed!
by Mark Campbell-Smith 09 Oct '09

09 Oct '09

Hi! I have just started to use dingaling again, and noticed I constantly get a stun error. 2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed! stun.fwdnet.net:3478 [Remote Address Error!] I have tried with stun.freeswitch.org and stun.fwdnet.net stun servers and keep getting this error with dingaling. I have no problems with inbound sip calls, so I don't think its the actual stun server. Has anyone else seen this? I am using: FreeSWITCH Version 1.0.trunk (14952) Thanks!

2 3

mod_fax compile fails
by Mark Campbell-Smith 09 Oct '09

09 Oct '09

HI all, I just tried to update to the latest svn and I get these errors right at the end after issuing a 'make current'. I am using Debian Lenny. making all mod_fax make[5]: Entering directory `/home/mark/freeswitch/src/mod/applications/mod_fax' make[6]: Entering directory `/home/mark/freeswitch/src/mod/applications/mod_fax' make[7]: Entering directory `/home/mark/freeswitch/libs/tiff-3.8.2' running /bin/sh ./configure --prefix=/usr/local/freeswitch --cache-file=/dev/null --srcdir=. --disable-shared --with-pic --no-create --no-recursion configure: error: cannot run /bin/sh config/config.sub make[7]: *** [config.status] Error 1 make[7]: Leaving directory `/home/mark/freeswitch/libs/tiff-3.8.2' make[6]: *** [../../../../libs/tiff-3.8.2/libtiff/libtiff.la] Error 2 make[6]: Leaving directory `/home/mark/freeswitch/src/mod/applications/mod_fax' make[5]: *** [all] Error 1 make[5]: Leaving directory `/home/mark/freeswitch/src/mod/applications/mod_fax' make[4]: *** [mod_fax-all] Error 1 make[4]: Leaving directory `/home/mark/freeswitch/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/home/mark/freeswitch/src' Making all in build make[3]: Entering directory `/home/mark/freeswitch/build' +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[3]: Leaving directory `/home/mark/freeswitch/build' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/home/mark/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/home/mark/freeswitch' make: *** [current] Error 2 Also, are the 'Leaving directory / all-recursive' errors going to cause a problem? Thanks! Any ideas what the cause is?

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linksys spa-3102 fax t38 + freeswitch media proxy
by Vladimir Elizarov 08 Oct '09

08 Oct '09

Hello. I want to set up faxing via the gateway linksys spa-3102 (with support for t38) via SIP. SIP-client -> linksys spa3102 -> fs -> provider <gateway name="callwithus"> .... <param name="inbound-late-negotiation" value="true"/> <param name="stun-enabled" value="true"/> </gateway> <extension name="Fax"> <condition field="destination_number" expression="^(30)(\d{11})$"> <action application="set" data="proxy_media=true"/> <action application="set" data="bypass_media=false"/> <action application="bridge" data="sofia/gateway/callwithus/*4087$2(a)sip.callwithus.com"/> </condition> </extension> I make a call, press start on the fax. Fax not sent. Get this in the log (UA freeswitch not trunk-12805): ------------------------------------------------------------------------ INVITE sip:*408774957558438@sip.callwithus.com SIP/2.0 Via: SIP/2.0/UDP 77.239.230.202:5080;rport;branch=z9hG4bK8yjrHeUprX2Hr Max-Forwards: 99 From: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN To: <sip:*408774957558438@sip.callwithus.com> Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 121302430 INVITE Contact: <sip:gw+callwithus@77.239.230.202:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 250 Remote-Party-ID: "Mihailova Ludmila" <sip:214@sip.callwithus.com>;party=calling;screen=yes;privacy=off v=0 o=- 216896551 216896551 IN IP4 192.168.50.51 s=- c=IN IP4 77.239.230.202 t=0 0 m=audio 32404 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 ------------------------------------------------------------------------ send 1333 bytes to udp/[38.99.70.232]:5060 at 15:05:34.485226: ------------------------------------------------------------------------ INVITE sip:*408774957558438@sip.callwithus.com SIP/2.0 Via: SIP/2.0/UDP 77.239.230.202:5080;rport;branch=z9hG4bK97BHK9BtN6r4K Max-Forwards: 99 From: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN To: <sip:*408774957558438@sip.callwithus.com> Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 121302431 INVITE Contact: <sip:gw+callwithus@77.239.230.202:5080;transport=udp> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 250 Remote-Party-ID: "Mihailova Ludmila" <sip:214@sip.callwithus.com>;party=calling;screen=yes;privacy=off v=0 o=- 216896551 216896551 IN IP4 192.168.50.51 s=- c=IN IP4 77.239.230.202 t=0 0 m=audio 32404 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 ------------------------------------------------------------------------ SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K From: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN To: <sip:*408774957558438@sip.callwithus.com> Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 121302431 INVITE Server: CWU SIP GW Content-Length: 0 ------------------------------------------------------------------------ recv 836 bytes from udp/[38.99.70.232]:5060 at 15:05:49.641312: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K Record-Route: <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6> From: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN To: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 121302431 INVITE User-Agent: CWU SIP-GW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:*408774957558438@204.74.213.1:5062> Content-Type: application/sdp Content-Length: 209 v=0 o=root 31797 31797 IN IP4 4.55.17.66 s=session c=IN IP4 4.55.17.66 t=0 0 m=audio 14104 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ------------------------------------------------------------------------ v=0 o=root 31797 31797 IN IP4 4.55.17.66 s=session c=IN IP4 4.55.17.66 t=0 0 m=audio 14104 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2009-10-06 19:03:31.112988 [NOTICE] sofia.c:3449 Pre-Answer sofia/external/*408774957558438(a)sip.callwithus.com! 2009-10-06 19:03:31.112988 [DEBUG] switch_channel.c:1822 Send signal sofia/internal/214(a)pbx0.tssec.lan [BREAK] 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2095 Set Codec sofia/external/*408774957558438(a)sip.callwithus.com PROXY/8000 20 ms 160 samples 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP [sofia/external/*408774957558438(a)sip.callwithus.com] 192.168.50.10:32404->4.55.17.66:14104 codec: 0 ms: 20 2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a timer 2009-10-06 19:03:31.112988 [DEBUG] switch_ivr_originate.c:2154 sofia/internal/214(a)pbx0.tssec.lan receive message [PROGRESS] 2009-10-06 19:03:31.112988 [INFO] switch_ivr_originate.c:2154 Sending early media 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:1322 sofia/internal/214(a)pbx0.tssec.lan Patched SDP --- v=0 o=root 31797 31797 IN IP4 4.55.17.66 s=session c=IN IP4 4.55.17.66 t=0 0 m=audio 14104 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 +++ v=0 o=root 31797 31797 IN IP4 4.55.17.66 s=session c=IN IP4 192.168.50.10 t=0 0 m=audio 32698 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP [sofia/internal/214(a)pbx0.tssec.lan] 192.168.50.10:32698->192.168.50.51:16474 codec: 0 ms: 30 2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a timer 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2508 Set comfort noise payload to 13 2009-10-06 19:03:31.112988 [DEBUG] sofia.c:3462 sofia/internal/214(a)pbx0.tssec.lan receive message [PROGRESS] 2009-10-06 19:03:31.116989 [NOTICE] mod_sofia.c:1521 Pre-Answer sofia/internal/214(a)pbx0.tssec.lan! 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/214(a)pbx0.tssec.lan [BREAK] 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_originate.c:2196 Originate Resulted in Success: [sofia/external/*408774957558438(a)sip.callwithus.com] 2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182 sofia/external/*408774957558438(a)sip.callwithus.com receive message [AUDIO_SYNC] 2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182 sofia/internal/214(a)pbx0.tssec.lan receive message [AUDIO_SYNC] 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:911 sofia/external/*408774957558438(a)sip.callwithus.com receive message [BRIDGE] 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal sofia/external/*408774957558438(a)sip.callwithus.com [BREAK] 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:918 sofia/internal/214(a)pbx0.tssec.lan receive message [BRIDGE] 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/214(a)pbx0.tssec.lan [BREAK] 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:962 (sofia/external/*408774957558438(a)sip.callwithus.com) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2009-10-06 19:03:31.116989 [DEBUG] sofia.c:3359 Channel sofia/internal/214(a)pbx0.tssec.lan entering state [early][183] 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/214(a)pbx0.tssec.lan [BREAK] 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:932 Send signal sofia/external/*408774957558438(a)sip.callwithus.com [BREAK] 2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:306 (sofia/external/*408774957558438(a)sip.callwithus.com) Running State Change CS_EXCHANGE_MEDIA 2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:343 (sofia/external/*408774957558438(a)sip.callwithus.com) State EXCHANGE_MEDIA 2009-10-06 19:03:31.116989 [DEBUG] mod_sofia.c:430 SOFIA LOOPBACK recv 822 bytes from udp/[38.99.70.232]:5060 at 15:05:58.503701: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K Record-Route: <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6> From: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN To: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 121302431 INVITE User-Agent: CWU SIP-GW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:*408774957558438@204.74.213.1:5062> Content-Type: application/sdp Content-Length: 209 v=0 o=root 31797 31798 IN IP4 4.55.17.66 s=session c=IN IP4 4.55.17.66 t=0 0 m=audio 14104 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ------------------------------------------------------------------------ send 762 bytes to udp/[38.99.70.232]:5060 at 15:05:58.507702: ------------------------------------------------------------------------ ACK sip:*408774957558438@204.74.213.1:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.10:5080;rport;branch=z9hG4bKaH59m4vXjFFQF Route: <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6> Max-Forwards: 70 From: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN To: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 121302431 ACK Contact: <sip:gw+callwithus@77.239.230.202:5080;transport=udp> Content-Length: 0 ------------------------------------------------------------------------ 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel sofia/external/*408774957558438(a)sip.callwithus.com entering state [ready][200] 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:1935 Send signal sofia/internal/214(a)pbx0.tssec.lan [BREAK] 2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3814 Channel [sofia/external/*408774957558438(a)sip.callwithus.com] has been answered 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182 sofia/external/*408774957558438(a)sip.callwithus.com receive message [AUDIO_SYNC] 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3819 sofia/internal/214(a)pbx0.tssec.lan receive message [ANSWER] 2009-10-06 19:03:39.975377 [DEBUG] sofia_glue.c:1322 sofia/internal/214(a)pbx0.tssec.lan Patched SDP --- v=0 o=root 31797 31797 IN IP4 4.55.17.66 s=session c=IN IP4 4.55.17.66 t=0 0 m=audio 14104 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 +++ v=0 o=root 31797 31797 IN IP4 4.55.17.66 s=session c=IN IP4 192.168.50.10 t=0 0 m=audio 32698 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2009-10-06 19:03:39.975377 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/214(a)pbx0.tssec.lan [BREAK] 2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3819 Channel [sofia/internal/214(a)pbx0.tssec.lan] has been answered 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182 sofia/internal/214(a)pbx0.tssec.lan receive message [AUDIO_SYNC] 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel sofia/internal/214(a)pbx0.tssec.lan entering state [completed][200] 2009-10-06 19:03:39.991381 [DEBUG] sofia.c:3359 Channel sofia/internal/214(a)pbx0.tssec.lan entering state [ready][200] recv 1054 bytes from udp/[38.99.70.232]:5060 at 15:06:05.577607: ------------------------------------------------------------------------ INVITE sip:gw+callwithus@192.168.50.10:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0 Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062 From: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 To: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN Contact: <sip:*408774957558438@204.74.213.1:5062> Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 102 INVITE User-Agent: CWU SIP-GW Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-info: SIP re-invite (T38 switchover) Content-Type: application/sdp Content-Length: 350 v=0 o=root 31797 31799 IN IP4 32.50.63.192 s=session c=IN IP4 32.50.63.192 t=0 0 m=image 14104 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:176 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy ------------------------------------------------------------------------ send 443 bytes to udp/[38.99.70.232]:5060 at 15:06:05.577607: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0 Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062 From: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 To: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805 Content-Length: 0 ------------------------------------------------------------------------ 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel sofia/external/*408774957558438(a)sip.callwithus.com entering state [received][100] 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3366 Remote SDP: v=0 o=root 31797 31799 IN IP4 32.50.63.192 s=session c=IN IP4 32.50.63.192 t=0 0 m=image 14104 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:176 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3683 Passing SDP to other leg. v=0 o=root 31797 31799 IN IP4 32.50.63.192 s=session c=IN IP4 32.50.63.192 t=0 0 m=image 14104 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:176 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3696 sofia/internal/214(a)pbx0.tssec.lan receive message [MEDIA_REDIRECT] 2009-10-06 19:03:47.049284 [DEBUG] mod_sofia.c:1195 sofia/internal/214(a)pbx0.tssec.lan Sending media re-direct: v=0 o=root 31797 31799 IN IP4 32.50.63.192 s=session c=IN IP4 32.50.63.192 t=0 0 m=image 14104 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:176 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy 2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1101 Remote address:port [192.168.50.51:16474] has not changed. 2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1322 sofia/internal/214(a)pbx0.tssec.lan Patched SDP --- v=0 o=root 31797 31799 IN IP4 32.50.63.192 s=session c=IN IP4 32.50.63.192 t=0 0 m=image 14104 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:176 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy +++ v=0 o=root 31797 31799 IN IP4 32.50.63.192 s=session c=IN IP4 192.168.50.10 t=0 0 m=image 32698 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:176 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy 2009-10-06 19:03:47.049284 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/214(a)pbx0.tssec.lan [BREAK] 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel sofia/internal/214(a)pbx0.tssec.lan entering state [calling][0] 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1101 Remote address:port [192.168.50.51:16474] has not changed. 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3158 Passing 200 OK to other leg 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3170 sofia/external/*408774957558438(a)sip.callwithus.com receive message [RESPOND] 2009-10-06 19:03:47.077291 [DEBUG] mod_sofia.c:1427 Responding with 200 [OK] 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1322 sofia/external/*408774957558438(a)sip.callwithus.com Patched SDP --- v=0 o=- 216899808 216899808 IN IP4 192.168.50.51 s=- c=IN IP4 192.168.50.51 t=0 0 m=image 16474 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy +++ v=0 o=- 216899808 216899808 IN IP4 192.168.50.51 s=- c=IN IP4 77.239.230.202 t=0 0 m=image 32404 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1110 AUDIO RTP CHANGING DEST TO: [32.50.63.192:14104] 2009-10-06 19:03:47.077291 [DEBUG] switch_core_session.c:630 Send signal sofia/external/*408774957558438(a)sip.callwithus.com [BREAK] 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel sofia/internal/214(a)pbx0.tssec.lan entering state [ready][200] 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3366 Remote SDP: v=0 o=- 216899808 216899808 IN IP4 192.168.50.51 s=- c=IN IP4 192.168.50.51 t=0 0 m=image 16474 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy send 1003 bytes to udp/[38.99.70.232]:5060 at 15:06:05.605615: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0 Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062 From: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 To: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 102 INVITE Contact: <sip:gw+callwithus@77.239.230.202:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 274 v=0 o=- 216896551 216896552 IN IP4 192.168.50.51 s=- c=IN IP4 77.239.230.202 t=0 0 m=image 32404 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy ------------------------------------------------------------------------ 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel sofia/external/*408774957558438(a)sip.callwithus.com entering state [completed][200] 2009-10-06 19:03:47.285347 [INFO] switch_rtp.c:1905 Auto Changing port from 32.50.63.192:14104 to 4.55.17.66:14104 recv 527 bytes from udp/[38.99.70.232]:5060 at 15:06:05.817672: ------------------------------------------------------------------------ ACK sip:gw+callwithus@192.168.50.10:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.2 Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK5ffd72cd;rport=5062 From: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 To: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN Contact: <sip:*408774957558438@204.74.213.1:5062> Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 102 ACK User-Agent: CWU SIP-GW Max-Forwards: 69 Content-Length: 0 ------------------------------------------------------------------------ 2009-10-06 19:03:47.289349 [DEBUG] sofia.c:3359 Channel sofia/external/*408774957558438(a)sip.callwithus.com entering state [ready][200] recv 476 bytes from udp/[38.99.70.232]:5060 at 15:06:47.588933: ------------------------------------------------------------------------ BYE sip:gw+callwithus@192.168.50.10:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0 Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062 From: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 To: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 103 BYE User-Agent: CWU SIP-GW Max-Forwards: 69 Content-Length: 0 ------------------------------------------------------------------------ 2009-10-06 19:04:29.060609 [NOTICE] sofia.c:328 Hangup sofia/external/*408774957558438(a)sip.callwithus.com [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-10-06 19:04:29.060609 [DEBUG] switch_channel.c:1726 Send signal sofia/external/*408774957558438(a)sip.callwithus.com [KILL] 2009-10-06 19:04:29.060609 [DEBUG] switch_core_session.c:932 Send signal sofia/external/*408774957558438(a)sip.callwithus.com [BREAK] 2009-10-06 19:04:29.060609 [DEBUG] switch_core_state_machine.c:437 thread mismatch skipping state handler. send 574 bytes to udp/[38.99.70.232]:5060 at 15:06:47.588933: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0 Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062 From: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 To: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 103 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------

6 8

freeswitch distributed
by Vladimir Elizarov 08 Oct '09

08 Oct '09

I want to make freeswitch distributed in such a situation. Several offices in each of fs. We need a unified dialplan, a single entrance/exit on the SIP external gateways, in the case of unavailability of one of the fs others work without his participation. How to understand. It should be the following. Base presence to keep in a separate database (pgpoll + pg). Dialplan, configure pull xml_curl. What can you tell from the web-application (wikipbx, fs2web etc)? The main task of which will impact dialplan, configure, directory?

2 1

How to limit the number of incoming+outgoing calls via specific gateway?
by Dmitry Bely 08 Oct '09

08 Oct '09

My SIP provider allows only one call (incoming or outgoing) via one SIP account. For FreeSWITCH I have configured it as public DID extension and outgoing gateway. Now I would like to transfer to another gw (or generate "limit exceded") when one tries to place an outgoing call while incoming call is in progress. How tho do that? Limiting the number of outgoing calls is easy (mod_limit), but how to take into account incoming one? - Dmitry Bely

6 10

Problem with gateway registration
by Nicolas Brenner 08 Oct '09

08 Oct '09

Hello everyone, I am trying to add a gateway, but after configuring it just like the others gateways I have, it is failing to register with a message like this: 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration Failed with status Operation has no matching challenge [904]. failure #1 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed Registration, setting retry to 10 seconds. I captured the sip traffic and noticed that when trying to register with one gateway (the one that works), I get a "Trying" reply immediately followed by a "401 Unauthorized" which contains a "WWW-Authenticate: digest" with a "qop=auth" parameter. Then Freeswitch replies with a second REGISTER including a large "Authorization: digest" section with cnonce and nc=00000001 parameters. The gateway which doesn't register, doesn't send the "qop=auth" parameter together with the "401 Unauthorized", and then Freeswitch sends a "Authorization: digest" section on the second REGISTER with no cnonce or nc parameters. I know very little abouth SIP, so I'm wondering what this "qop=auth" parameter means and how does it affect the registration process. Is there any way to do without the qop=auth parameter? Also, I tried registering with X-Lite directly to the gateway, and it worked, so it appears to be a problem in the Freeswitch/gateway combination. (Note: X-Lite sends an "Authorization: digest" section on the _first_ REGISTER, apparently this makes a difference) Attached is a sip trace for the registration traffic when doing "sofia profile external restart reloadxml" on the cli, captured with "tshark -i eth0 -o "rtp.heuristic_rtp: TRUE" -w /tmp/capture.pcap -b filesize:51200 -b files:100 -R 'sip or rtp or icmp or dns or rtcp or t38'" Thanks! Nicolas

5 18

unable to configure Digium TDM400P
by Woody Dickson 08 Oct '09

08 Oct '09

Hi, I am trying to setup a Digium TDM400P following the instruction on the wiki. It is a 1 fxo and 1 fxs card, so I tried loadzone=in defaultzone=in fxsks=2 fxoks=1 and loadzone=in defaultzone=in fxsks=1 fxoks=2 None works. Does anyone know how it should be configured? Here is what I get by following the wiki. [root@localhost zaptel]# ztcfg -vv Zaptel Version: SVN-branch-1.4-r4629M Echo Canceller: MG2 Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) 2 channels to configure. Changing signalling on channel 1 from FXO Kewlstart to FXS Kewlstart Changing signalling on channel 2 from FXS Kewlstart to FXO Kewlstart [root@localhost zaptel]# ztcfg -vv Zaptel Version: SVN-branch-1.4-r4629M Echo Canceller: MG2 Configuration ====================== Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels to configure. Changing signalling on channel 1 from FXS Kewlstart to FXO Kewlstart Changing signalling on channel 2 from FXO Kewlstart to FXS Kewlstart [root@localhost zaptel]# lspci 00:14.0 Ethernet controller: Digium, Inc. TDM400P (rev 11)

2 3