June 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Location where FreeSWITCH mod_callcenter read audio files
by Joel White 11 Jun '14

11 Jun '14

I have made some recordings for the callcenter. I have placed them in several locations (sounds, recordings) yet I cannot get the 'announce-sound' option in mod_callcenter to play these files. Can someone help me in locating the directory that the files need to be placed for the system to be able to access them? Thank you in advance

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ClueCon Weekly! June 11 - FreeSWITCH's fs_cli tricks!
by Ken Rice 11 Jun '14

11 Jun '14

Hey Guys Join us today for fs_cli tricks! You know it and you use it all the time! But when¹s the last time you looked at some of the tricks you can do with fs_cli! Join us today at 1PM EST on sip:888@conference.freeswitch.org to see what fun tricks you can do and help save time! Ken -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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ClueCon Speakers Announced!
by Ken Rice 11 Jun '14

11 Jun '14

Hey ClueCon¹ers! If you havent visited ClueCon.com lately Now might be a Good time to do so! We have added a ton of confirmed speakers and you can see them at https://cluecom.com/speakers !! Come for ClueCon Stay for the Training! Space is limited in the training so register today! The course will include these topics: · Quick-start on FreeSWITCH Basics · Setting up and using WebRTC · Security (SRTP and ZRTP) and Certificates · Using FreeSWITCH's "other" configs - FreeSWITCH as a proxy or hybrid · Multi-tenancy with FreeSWITCH · XML/Curl and controlling FreeSWITCH real time · Raw Event Sockets Who should attend: · VoIP providers seeking ³Open Source² platforms to enhance their businesses · Anyone seeking proficiency in FreeSWITCH · Network Consultants and VARs who need a jump start in the technology · Developers who want to use FreeSWITCH to create new telephony applications and appliances · Savvy Tech People who like to learn and build cool stuff! Schedule Details: · The course will begin at 9am a the (PLEASE NAME THE LOCATION) · The course will end at around 4pm with a one-hour slot to play and answer questions where the teacher will remain present (as long as you have questions!) -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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mod_skypopen and mod_sms
by Russell Treleaven 10 Jun '14

10 Jun '14

It is possible but you will need to tweak headers in the chatplan. first step though is <param name="report_incoming_chatmessages" value="true"/> Sorry I can't give more specifics at the moment. On Tue, Jun 10, 2014 at 7:08 AM, Mark Campbell-Smith < mcampbellsmith(a)gmail.com> wrote: > 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 > > _________________________________________________________________________ > 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 > >

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STUN Binding request times out after 200 OK received on client
by Jayesh Nambiar 10 Jun '14

10 Jun '14

Hi, I am developing an iOS calling App with libnice for ICE capabilities. The calls go through freeswitch and since freeswitch supports ICE, I expected it to work out of the box. But I have a problem. The way libnice works is as follows: 1) The client sends INVITE with his candidates. 2) Freeswitch parses the candidates. 3) Immediately after 200 OK is received, I can see FreeSWITCH choosing an ICE candidate and logging "Activating Audio ICE" indicating the IP and Port and it sends media/RTP to. 4) But the libnice agent tries to look at the candidates received in 200 OK and tries to send a STUN binding request to FreeSWITCH IP as that IP is mentioned as ICE candidate in the 200 OK coming from freeswitch. 5) The FreeSWITCH doesn't respond to that STUN binding request and libnice agent times out. 6) Ideally, if libnice gets back a response from the freeswitch, it should initiate a Re-INVITE with the candidate pair selected. Now here, the freeswitch starts sending RTP to client's candidate, but the client gets stuck because the candidate pair is not formed as the libnice agent keeps waiting for a response for the STUN Binding Request. Is there anything that I am missing here. Any help is highly appreciated. Thanks, --- Jayesh

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httapi and getting bridge hangup cause
by Colin Martin 10 Jun '14

10 Jun '14

Hi, I’m new to Freeswitch, and gradually getting to grips with the httapi application. I’m having trouble getting the hangup cause back after a bridge is complete. I’ve tried Googling around the issue, but I can’t find anything that would explain why this doesn’t work. So, the web server initiates the bridge with the following: <?xml version="1.0" encoding="UTF-8" standalone="no"?><document type="text/freeswitch-httapi"><work><execute application="set">call_timeout=30</execute><execute application="bridge">sofia/gateway/pbx/200(a)destination.com</execute></work></document> The call is placed and the bridge works as expected. The called party then hangs up. As expected, the httapi application makes a further request, but only sends a small subset of the query parameters that it would usually send, so I can’t identify the hangup cause. I expect one of my assumptions is wrong and this is expected behaviour. What am I doing wrong, and how should I be getting the hangup cause? Thanks for any help you can provide! Colin Martin

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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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