June 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Error in codec negotiation: What are all the places that specify accepted codecs for outgoing calls?
by Mark Boots 10 Jun '14

10 Jun '14

Thanks for the help. I'm working on a clean log. We had many outgoing calls at the same time, the console log is mixed up with other calls. Digging more into our logs, it seems the issue is sporadic: 50% of the time we can connect to a switch offering only PCMA, and other times we only offer PCMU and get this codec negotiation error. The configuration has not changed across calls: >> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM"/> >> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> I'll work on a console log and sip trace. In the meantime, any idea what could cause variation between calls in the codecs offered? +Mark > From: Kristian Kielhofner <kris(a)kriskinc.com> > Subject: Re: [Freeswitch-users] Error in codec negotiation: What are all the places that specify accepted codecs for outgoing calls? > Date: 6 June 2014 18:17:47 GMT > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > We would need to see a full siptrace and console log to see what's > happening here. > > On Fri, Jun 6, 2014 at 10:27 AM, Mark Boots <mark(a)votomobile.org> wrote: >> We're using a major VoIP terminator and finding that on many outgoing calls, >> Freeswitch hangs up with INCOMPATIBLE_DESTINATION after trying to negotiate >> codecs. >> >> The carrier offers PCMA, and it looks like Freeswitch refuses to accept >> anything but PCMU: >> >> >> 2014-06-06 13:18:54.115197 [NOTICE] sofia_media.c:92 Pre-Answer >> sofia/external/0011104555493618652! >> 2014-06-06 13:18:54.115197 [DEBUG] switch_channel.c:3277 >> (sofia/external/0011104555493618652) Callstate Change DOWN -> EARLY >> 2014-06-06 13:18:54.175210 [DEBUG] switch_core_session.c:1005 Send signal >> sofia/external/0011104555391494140 [BREAK] >> 2014-06-06 13:18:54.175210 [DEBUG] switch_core_session.c:1005 Send signal >> sofia/external/0011104555391494140 [BREAK] >> 2014-06-06 13:18:54.175210 [DEBUG] sofia.c:5741 Channel >> sofia/external/0011104555391494140 entering state [proceeding][183] >> 2014-06-06 13:18:54.175210 [DEBUG] sofia.c:5750 Remote SDP: >> v=0 >> o=- 313783328916230866 1 IN IP4 81.52.176.165 >> s=- >> c=IN IP4 81.52.176.165 >> t=0 0 >> a=sendrecv >> m=audio 25080 RTP/AVP 8 13 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:13 CN/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> >> 2014-06-06 13:18:54.175210 [DEBUG] switch_core_media.c:2585 Set comfort >> noise payload to 13 >> 2014-06-06 13:18:54.175210 [DEBUG] switch_core_media.c:2662 Audio Codec >> Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] >> 2014-06-06 13:18:54.175210 [DEBUG] switch_core_media.c:2662 Audio Codec >> Compare [CN:13:8000:20:0]/[PCMU:0:8000:20:64000] >> 2014-06-06 13:18:54.175210 [DEBUG] switch_core_media.c:2662 Audio Codec >> Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] >> 2014-06-06 13:18:54.175210 [DEBUG] switch_core_media.c:2824 Set 2833 dtmf >> send payload to 101 >> 2014-06-06 13:18:54.175210 [DEBUG] switch_core_session.c:1005 Send signal >> sofia/external/0011104555391494140 [BREAK] >> 2014-06-06 13:18:54.175210 [NOTICE] sofia.c:5868 Hangup >> sofia/external/0011104555391494140 [CS_CONSUME_MEDIA] >> [INCOMPATIBLE_DESTINATION] >> >> >> However, our codec settings in vars.xml specify: >> >> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM"/> >> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> >> >> The sip profile uses these settings. Any idea how to debug what codecs >> Freeswitch is willing to accept? >> >> Thanks! >> +Mark >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> 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 >> > > > > -- > Kristian Kielhofner

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Freeswitch with Presence, and databases
by Yehavi Bourvine 10 Jun '14

10 Jun '14

Hello, I see here from time to time a suggestion to turn off presence in order to free CPU cycles. We are using Freeswitch inside an organization environment, which means that presence is crucial. I am also using the limit function using ODBC backend. Currently we have 300 extensions, but I would like to be prepared for an expansion (over 1,000), Hence, I have a few questions: - I currently have presence enabled for all phones; shall I segragate the phones into two groups, one with presence, one without? - I am using MySQL via ODBC interface for the core and our usage. It is on a "regular" disk, and I see no performance problems at present. Shall I consider moving to Postgress, or continue with MySQL? - All the phones share the same SIP profile. Will this be a bottleneck? The audio is passed through FS (in order to detect fax and DTMFs). Thanks! __Yehavi:

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WebRTC does not seem to work with FreeSWITCH on ChromeOS 36 beta
by Oleg Stolyar 10 Jun '14

10 Jun '14

Hi FreeSWITCH team, it seems that WebRTC on ChromeOS 36 beta does not work with FreeSWITCH. I tried to go to https://webrtc.freeswitch.org/sipjs/index.html and with default settings call into ext 9664 to listen to hold music. It works fine on Windows and Mac all versions of Chrome and on ChromeOS versions 34 and 35 but does not seem to work on ChromeOS beta 36. I am not sure why. I am using JsSip client in my own app and see the same problem there. I don't think the problem is with JsSip or its offshot sipjs which webrtc.freeswitch.org uses since jssip's own demo seems to work fine http://tryit.jssip.net/

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rewriting sip contact field
by ik 10 Jun '14

10 Jun '14

Hello, I have a network with two local net's: A. 10.0.0.0/16 B. 172.16.0.0./12 . The server where freeswitch is located, has only a 10.0.0.2 ethernet device, and no other devices. I am able to talk with a server that is over the 172.16.0.0./12 address, that does not know about the 10 address segments. The problem is that the contact field contains only the 10.0.0.2 address, rather a different address that is required by the 172 address set (172.20.12.30), although it is routed to the freeswitch machine, so it place it on the SIP contact field, and the other side disprove of it and return me 401 message. I can not change this fucked up networking issue, or create even a virtual device on freeswitch server that it could bind itself to a 172 address. The really weird part, is that the sdp actually does send the 172.20.12.30 address, but the contact is 10.0.0.2. Is there a way in freeswitch to rewrite the rule to set the address to 172 instead ? Thanks, Ido

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mod_skypopen and mod_sms
by Mark Campbell-Smith 10 Jun '14

10 Jun '14

Hi! I read in Jira that it should be possible to use mod_sms with chat messages received from mod_skypopen. How do I do this? When I send a message to FS, I see the debug messages received from mod_skypopen, but do not see mod_sms processing the message. 2014-06-10 21:02:48.956151 [DEBUG] skypopen_protocol.c:207 [1.5.13b+git~20140603T193316Z~70b03acf71~64bit ] [DEBUG_SKYPE 207 ][skype101 ][IDLE,IDLE] READING: |||CHAT #skype1/$skyp2;f067e42dadeb74b ACTIVITY_TIMESTAMP 1402398167||| 2014-06-10 21:02:48.956151 [DEBUG] skypopen_protocol.c:207 [1.5.13b+git~20140603T193316Z~70b03acf71~64bit ] [DEBUG_SKYPE 207 ][skype101 ][IDLE,IDLE] READING: |||CHATMESSAGE 1453 STATUS RECEIVED||| I can get normal SIP messages to work. Any ideas? Thanks

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Freeswitch and message queue
by Oscar Franzen 10 Jun '14

10 Jun '14

Hi! Does anyone have any experience with having freeswitch emit all channel/call events to a message queue (e.x. RabbitMQ)? -- Regards *Oscar Franzén**Developer* *Advisa* oscar.franzen(a)advisa.se Mobil: 070-720 45 81 Besöksadress: Kungsgatan 30, Stockholm www.advisa.se ––––––––––––––––––––––––––– Informationen i detta meddelande är konfidentiell och endast avsedd för den/de angivna mottagarna. Om Du inte är en avsedd mottagare, ber vi Dig att inte läsa, kopiera eller använda informationen i meddelandet samt att inte heller vidaresända informationen eller delge den till annan. Vi ber Dig även att omedelbart meddela avsändaren att Du erhållit meddelandet felaktigt samt att därefter genast radera det från Ditt system. Advisa ansvarar inte för eventuella personliga åsikter eller värderingar i detta meddelande.

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INCOMPATIBLE_DESTINATION
by Shlomi Agiv 10 Jun '14

10 Jun '14

Hi, I'm running freswitch version 1.4.4 and getting a strange codec error when i call from a phone running CSIPSIMPLE(registered as user 1000) to 1005, the call hangs up and I get: *2014-06-05 08:31:59.168784 [INFO] switch_ivr_originate.c:1191 Sending early media* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [SILK:99:24000:20:0]/[G722:9:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [SILK:99:24000:20:0]/[PCMU:0:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [SILK:99:24000:20:0]/[PCMA:8:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [SILK:99:24000:20:0]/[GSM:3:8000:20:13200]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:30:32000]/[G722:9:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:30:32000]/[PCMU:0:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:30:32000]/[PCMA:8:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:30:32000]/[GSM:3:8000:20:13200]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMA:8:8000:30:32000]/[G722:9:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMA:8:8000:30:32000]/[PCMU:0:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMA:8:8000:30:32000]/[PCMA:8:8000:20:64000]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMA:8:8000:30:32000]/[GSM:3:8000:20:13200]* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3309 Set telephone-event payload to 101* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_media.c:3626 Set 2833 dtmf send/recv payload to 101* *2014-06-05 08:31:59.168784 [ERR] mod_sofia.c:2201 CODEC NEGOTIATION ERROR. SDP:* *v=0* *o=- 3610945918 3610945918 IN IP4 10.0.1.46* *s=pjmedia* *c=IN IP4 10.0.1.46* *t=0 0* *m=audio 4020 RTP/AVP 99 0 8 101* *c=IN IP4 10.0.1.46* *a=rtpmap:99 SILK/24000* *a=fmtp:99 useinbandfec=0* *a=rtpmap:0 PCMU/8000* *a=rtpmap:8 PCMA/8000* *a=rtpmap:101 telephone-event/8000* *a=fmtp:101 0-15* *a=rtcp:4021 IN IP4 10.0.1.46* *2014-06-05 08:31:59.168784 [DEBUG] switch_core_session.c:907 Send signal sofia/internal/1000(a)10.0.1.233 <1000(a)10.0.1.233> [BREAK]* *2014-06-05 08:31:59.168784 [NOTICE] switch_channel.c:3432 Hangup sofia/internal/1000(a)10.0.1.233 <1000(a)10.0.1.233> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]* when i call using cmd 'originate user/1005 &bridge(user/1000)' The call goes through and I get: *2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]* *2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3437 Audio Codec Compare [PCMU:0:8000:20:64000] ++++ is saved as a match* *2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]* *2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMA:8:8000:20:64000]* *2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3383 Audio Codec Compare [PCMU:0:8000:20:64000]/[GSM:3:8000:20:13200]* *2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:3309 Set telephone-event payload to 101* *2014-06-05 08:25:18.073562 [DEBUG] switch_core_media.c:2343 Set Codec sofia/internal/sip:1000@10.0.1.46:51113 <http://sip:1000@10.0.1.46:51113> PCMU/8000 20 ms 160 samples 64000 bits* *2014-06-05 08:25:18.073562 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:1000@10.0.1.46:51113 <http://sip:1000@10.0.1.46:51113> Original read codec set to PCMU:0* *2014-06-05 08:25:18.093570 [DEBUG] switch_core_media.c:3617 Set 2833 dtmf send payload to 101* *2014-06-05 08:25:18.093570 [DEBUG] switch_core_media.c:4841 AUDIO RTP [sofia/internal/sip:1000@10.0.1.46:51113 <http://sip:1000@10.0.1.46:51113>] 10.0.1.233 port 32636 -> 10.0.1.46 port 4016 codec: 0 ms: 20* *2014-06-05 08:25:18.093570 [DEBUG] switch_rtp.c:3349 Starting timer [soft] 160 bytes per 20ms* *2014-06-05 08:25:18.113587 [INFO] switch_core_media.c:5053 Activating RTCP PORT 4017* Even stranger, when I call using the same configuration under 1.2.17, the call goes through: *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [SILK:99:24000:20:0]/[G722:9:8000:20:64000]* *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [SILK:99:24000:20:0]/[PCMU:0:8000:20:64000]* *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [SILK:99:24000:20:0]/[PCMA:8:8000:20:64000]* *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [SILK:99:24000:20:0]/[GSM:3:8000:20:13200]* *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]* *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5278 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]* *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:3186 Set Codec sofia/internal/1000(a)10.0.1.233 <1000(a)10.0.1.233> PCMU/8000 20 ms 160 samples 64000 bits* *2014-06-05 08:43:39.400774 [DEBUG] switch_core_codec.c:111 sofia/internal/1000(a)10.0.1.233 <1000(a)10.0.1.233> Original read codec set to PCMU:0* *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:5446 Set 2833 dtmf send/recv payload to 101* *2014-06-05 08:43:39.400774 [DEBUG] sofia_glue.c:3445 AUDIO RTP [sofia/internal/1000(a)10.0.1.233 <1000(a)10.0.1.233>] 10.0.1.233 port 23172 -> 10.0.1.46 port 4004 codec: 0 ms: 20* what could possibly be the reason for that? I have little understanding of codecs handling. Thanks, Shlomi Agiv

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bind_digit_action targeting/responding on opposite leg
by Dmitriy Shumaev 10 Jun '14

10 Jun '14

Hi. I want to use bind_digit_action instead of bind_meta_app. There is one problem - I need to do actions on opposite leg. Using bind_meta_app I can do something like this: <action application="bind_meta_app" data="2 ab o execute_extension::INFO_calloverreq XML default"/>  , but how can I do the same using bind_digit_action - ?. With best regards, Shumaev D.A., programmer @ KBR Ltd.

2 2

Unable to call extension
by Richard Mace 09 Jun '14

09 Jun '14

Hi All, I have a vanilla debian 7 FreeSWITCH installation with a file called 201.xml in /etc/freeswitch/dialplan/default/ When I dial 201 (from extension 203) it won't ring the extension. I have included a SIP trace on: www.rocksolidpbx.co.uk/download/sofia_sip_trace.txt Any ideas what I'm doing wrong? Thanks, as always, in advance Richard

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Getting logs of registrations/deregistrations
by Alex Lake 09 Jun '14

09 Jun '14

Is there a way to get the console to show end-user registrations and de-registrations as they happen? I note that registrations are OK, you get something like this: 2014-06-09 13:37:44.091960 [WARNING] sofia_reg.c:1541 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1129301(a)004-1129.sb12.dmclub.org] from ip 84.92.90.149 But de-registrations are very quiet. Also, is there an api command to send a "SIP OPTIONS" message to a particular registered session?

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