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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

Given that our provider does this 40% of the time in multiple countries, is there anything we can do to get this working with Freeswitch? It seems logical for this to work: we offered a range of codecs, the other end chose one, then for another part of the call chose another. Both codecs are supported by our end, so why should we reject the second just because another was used first? According to the console, Freeswitch is doing a codec renegotiation process... It's just not accepting anything other than 1 codec at this point. +Mark From: Michael Jerris <mike(a)jerris.com> Subject: Re: [Freeswitch-users] Error in codec negotiation: What are all the places that specify accepted codecs for outgoing calls? Date: 10 June 2014 12:42:31 GMT To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Hmm, not sure thats true. You are not supposed to change codecs before the dialog is complete like this, its a violation of sdp o/a. On Jun 10, 2014, at 4:50 AM, Kristian Kielhofner <kris(a)kriskinc.com> wrote: > Hi Mark, > > Thanks for providing the full log. > > While goofy this is perfectly valid behavior. Because your instance of Freeswitch offers pcma and pcmu it's ok for the other side to switch between the two in 183. > > A few things to try: > > - Limit your outbound offers to pcma or pcmu only (one codec). > > - The remote side appears to support 100rel (PRACK). Try enabling it in Freeswitch. While a long shot it's possible that PRACK'ing the initial received 183 might change the codec swap behavior of the remote end. > > - Try Freeswitch 1.4 or master. There's been some significant changes in media/SDP behavior, particularly when the remote end does technically standards compliant but goofy things with multiple codec offers. > > If none of these works you're only option may be to have the carrier take some action. > > On Monday, June 9, 2014, Mark Boots <mark(a)votomobile.org> wrote: > Hi Kristian, > > I've narrowed this INCOMPATIBLE_DESTINATION error down to a specific case: > > 1 - Freeswitch sends the outgoing call, and offers PCMU/PCMA/GSM > 2 - Other end accepts only PCMU > 3 - At some point after Pre-answer, the other end issues a 183 Session Progress, changing to only accept PCMA. > 4 - Freeswitch attempts to select new codecs but only looks at PCMU, and cancels the call with INCOMPATIBLE_DESTINATION. > > > Here is the console log and sip trace... Hope it helps! > > +Mark > > 2014-06-09 21:35:39.615207 [DEBUG] switch_ivr_originate.c:2069 Parsing global variables > 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_request_uuid]=[004fd8a2-f01e-11e3-844d-22000ad937f0] > 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_answer_url]=[http://10.8.0.1/deliverylogs/answer/752471] > 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_ring_url]=[http://127.0.0.1/callqueue/ring.php] > 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_hangup_url]=[http://127.0.0.1/callqueue/hangup.php] > 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [origination_caller_id_number]=[12026004299] > 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_from]=[12026004299] > 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_to]=[0011104555592211972] > 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_app]=[true] > 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [originate_timeout]=[60] > 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [ignore_early_media]=[true] > 2014-06-09 21:35:39.615207 [NOTICE] switch_channel.c:1049 New Channel sofia/external/0011104555592211972 [005052e6-f01e-11e3-bbae-cbf0f4eac998] > 2014-06-09 21:35:39.615207 [DEBUG] mod_sofia.c:5237 (sofia/external/0011104555592211972) State Change CS_NEW -> CS_INIT > 2014-06-09 21:35:39.615207 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/0011104555592211972 [BREAK] > 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:418 (sofia/external/0011104555592211972) Running State Change CS_INIT > 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:458 (sofia/external/0011104555592211972) State INIT > 2014-06-09 21:35:39.615207 [DEBUG] mod_sofia.c:87 sofia/external/0011104555592211972 SOFIA INIT > 2014-06-09 21:35:39.615207 [DEBUG] sofia_glue.c:2746 sofia/external/0011104555592211972 sending invite version: 1.2.24 git 7b74ee3 2014-06-03 00:26:24Z 32bit > Local SDP: > v=0 > o=FreeSWITCH 1402325141 1402325142 IN IP4 54.203.245.50 > s=FreeSWITCH > c=IN IP4 54.203.245.50 > t=0 0 > m=audio 24598 RTP/AVP 0 8 3 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > 2014-06-09 21:35:39.615207 [DEBUG] mod_sofia.c:127 (sofia/external/0011104555592211972) State Change CS_INIT -> CS_ROUTING > 2014-06-09 21:35:39.615207 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/0011104555592211972 [BREAK] > 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:458 (sofia/external/0011104555592211972) State INIT going to sleep > send 1071 bytes to udp/[108.59.2.133]:5060 at 21:35:39.620673: > ------------------------------------------------------------------------ > INVITE sip:0011104555592211972@108.59.2.133 SIP/2.0 > Via: SIP/2.0/UDP 54.203.245.50:5080;rport;branch=z9hG4bKUNeNtZcZp5ZmB > Max-Forwards: 70 > From: "" <sip:12026004299@54.203.245.50>;tag=9Z1HtD145Qv3K > To: <sip:0011104555592211972@108.59.2.133> > Call-ID: d7aeb7a1-6ac0-1232-718c-22000ad937f0 > CSeq: 60838805 INVITE > Contact: <sip:gw+voxbeam_outbound@54.203.245.50:5080;transport=udp;gw=voxbeam_outbound> > User-Agent: FreeSWITCH-mod_sofia/1.2.24+git~20140603T002624Z~7b74ee3955~32bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 207 > X-FS-Support: update_display,send_info > Remote-Party-ID: <sip:12026004299@54.203.245.50>;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1402325141 1402325142 IN IP4 54.203.245.50 > s=FreeSWITCH > c=IN IP4 54.203.245.50 > t=0 0 > m=audio 24598 RTP/AVP 0 8 3 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > ------------------------------------------------------------------------ > 2014-06-09 21:35:39.615207 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/0011104555592211972 [BREAK] > 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:418 (sofia/external/0011104555592211972) Running State Change CS_ROUTING > 2014-06-09 21:35:39.615207 [DEBUG] sofia.c:5845 Channel sofia/external/0011104555592211972 entering state [calling][0] > 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:474 (sofia/external/0011104555592211972) State ROUTING > 2014-06-09 21:35:39.615207 [DEBUG] mod_sofia.c:150 sofia/external/0011104555592211972 SOFIA ROUTING > 2014-06-09 21:35:39.615207 [DEBUG] switch_ivr_originate.c:67 (sofia/external/0011104555592211972) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2014-06-09 21:35:39.615207 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/0011104555592211972 [BREAK] > 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:474 (sofia/external/0011104555592211972) State ROUTING going to sleep > 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:418 (sofia/external/0011104555592211972) Running State Change CS_CONSUME_MEDIA > 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:493 (sofia/external/0011104555592211972) State CONSUME_MEDIA > 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:493 (sofia/external/0011104555592211972) State CONSUME_MEDIA going to sleep > recv 367 bytes from udp/[108.59.2.133]:5060 at 21:35:39.725204: > ------------------------------------------------------------------------ > SIP/2.0 100 Giving a try > Via: SIP/2.0/UDP 54.203.245.50:5080;received=54.203.245.50;rport=5080;branch=z9hG4bKUNeNtZcZp5ZmB > From: "" <sip:12026004299@54.203.245.50>;tag=9Z1HtD145Qv3K > To: <sip:0011104555592211972@108.59.2.133> > Call-ID: d7aeb7a1-6ac0-1232-718c-22000ad937f0 > CSeq: 60838805 INVITE > Server: OpenSIPS (1.8.2-notls (x86_64/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 657 bytes from udp/[108.59.2.133]:5060 at 21:35:41.191275: > ------------------------------------------------------------------------ > SIP/2.0 183 Session Progress > Via: SIP/2.0/UDP 54.203.245.50:5080;rport;branch=z9hG4bKUNeNtZcZp5ZmB > From: "" <sip:12026004299@54.203.245.50>;tag=9Z1HtD145Qv3K > To: <sip:0011104555592211972@108.59.2.133>;tag=9312047651043624688 > Call-ID: d7aeb7a1-6ac0-1232-718c-22000ad937f0 > CSeq: 60838805 INVITE > Content-Type: application/sdp > Contact: <sip:callee@108.59.2.133;did=795.65454242> > Supported: timer,100rel > Content-Length: 226 > > v=0 > o=RISNEXT02 869 1999 IN IP4 80.84.30.41 > s=sip call > c=IN IP4 80.84.30.41 > t=0 0 > m=audio 42108 RTP/AVP 0 13 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:13 CN/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > ------------------------------------------------------------------------ > 2014-06-09 21:35:41.175213 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/0011104555592211972 [BREAK] > 2014-06-09 21:35:41.175213 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/0011104555592211972 [BREAK] > 2014-06-09 21:35:41.175213 [DEBUG] sofia.c:5845 Channel sofia/external/0011104555592211972 entering state [proceeding][183] > 2014-06-09 21:35:41.175213 [DEBUG] sofia.c:5858 Remote SDP: > v=0 > o=RISNEXT02 869 1999 IN IP4 80.84.30.41 > s=sip call > c=IN IP4 80.84.30.41 > t=0 0 > m=audio 42108 RTP/AVP 0 13 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:13 CN/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > > 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3192 Set Codec sofia/external/0011104555592211972 PCMU/8000 20 ms 160 samples 64000 bits > 2014-06-09 21:35:41.175213 [DEBUG] switch_core_codec.c:111 sofia/external/0011104555592211972 Original read codec set to PCMU:0 > 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:5444 Set 2833 dtmf send payload to 101 > 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3451 AUDIO RTP [sofia/external/0011104555592211972] 10.217.55.240 port 24598 -> 80.84.30.41 port 42108 codec: 0 ms: 20 > 2014-06-09 21:35:41.175213 [DEBUG] switch_rtp.c:2040 Starting timer [soft] 160 bytes per 20ms > 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3718 Set 2833 dtmf send payload to 101 > 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3724 Set 2833 dtmf receive payload to 101 > 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3751 sofia/external/0011104555592211972 Set rtp dtmf delay to 40 > 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3757 Set comfort noise payload to 13 > 2014-06-09 21:35:41.175213 [NOTICE] sofia_glue.c:4362 Pre-Answer sofia/external/0011104555592211972! > _______________________________________________ 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

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performance hints
by Vladimir Getmanshchuk 10 Jun '14

10 Jun '14

I've finally reached announced performance on my FS. 1. Moved FS from KVM virtualization to bare metal 2. <param name="rtp-autofix-timing" value="false"/> at sofia.conf.xml -- Yours sincerely, Vladimir Getmanshchuk

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Passing through early media
by Alex Balashov 10 Jun '14

10 Jun '14

Hello, I've got a shockingly basic question that I can't seem to figure out the answer to: I have Leg A (handset) that calls Freeswitch, and the call is bridged to leg B, on which an announcement is played (183) and then a 503 is returned. I can't for the life of me get the announcement to play on leg A; it just gives ordinary ringback. I've tried with and without various regional ring options, and most importantly, tried explicitly setting ignore_early_media: <action application="bridge" data="{ignore_early_media=false}sofia/${use_profile}/1$1(a)xxx.xxx.xxx.xxx"/> ... albeit without success. Freeswitch still imposes its own ringback. Any pointers would be appreciated! -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

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Free G729
by Pezhman Lali 10 Jun '14

10 Jun '14

Dear, I am looking for a .so to have g729 (transcoding) in freeswitch as free. There are some docs linked to asterisk.hosting.lv where good place to get codec_g729.so for asterisk, not for freeswitch. for example: http://www.opentelecoms.org/codecs-open-source-g729-g723.1-asterisk but it does not work with freeswitch, ( a lot of undefined symbols ...). Could you please let me know if any free mod_g729.so is available. Best --

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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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