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Hi Reg IVR and codec negotiation.
by indra sena 08 Jan '15

08 Jan '15

Hi , Happy new year to every body. I have observed in freeswitch that, In case of IVR scenario prior to the bridge FreeSwitch plays IVR with the 1st priority codec then invite to the bridge endpoint (B-leg) with SDP containing only the codec negotiated for IVR. Also the answer (183 ) to originator(A-leg) with SDP containing only with IVR negotiated codec. I am having one issue here for example. Originator(A-leg) sends invite with G729, G711 and Freeswitch negotiated with G729 and starts playing IVR with G729 and answered(183) to originator with SDP containing only G729. And Invited to termination endpoint with SDP having only G729 and termination gateway having support of only G711, and it is rejecting call after receiving invite with only G729. It should be work like after IVR , invite can be having sdp with all the originator supported codecs and if termination (b-leg) having different priority codecs then it can send re-Invite sanding the same. I have observed one issue in freeswitch Jeera ( https://freeswitch.org/jira/browse/FS-880) , but there is no solution in that page. Do we have any solution for this right now ? Your answers will be very much helpful for me. I appreciate if you can quick response or give some solution for this. Thanks in advance. Thanks & Regards, Indra..

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BLF on Cisco 303/504/525
by Paul Nebb 08 Jan '15

08 Jan '15

Does anyone have templates or articles that would assist in setting up BLF on Cisco 303/504/525 Paul

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Multiple P-Asserted-Identity Headers
by George F. Phelps 07 Jan '15

07 Jan '15

How do I create two (2) unique, P-Asserted-Identity headers? As required by RFC 3325 <http://www.ietf.org/rfc/rfc3325.txt> : A P-Asserted-Identity header field value MUST consist of exactly one name-addr or addr-spec. There may be one or two P-Asserted-Identity values. If there is one value, it MUST be a sip, sips, or tel URI. If there are two values, one value MUST be a sip or sips URI and the other MUST be a tel URI. A valid, multiple P-Asserted-Identity headers example, taken from RFC 3325: INVITE sip:+14085551212@proxy.pstn.net SIP/2.0 Via: SIP/2.0/TCP useragent.cisco.com;branch=z9hG4bK-124 Via: SIP/2.0/TCP proxy.cisco.com;branch=z9hG4bK-abc To: <sip:+14085551212@cisco.com> From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=9802748 Call-ID: 245780247857024504 CSeq: 2 INVITE Max-Forwards: 69 P-Asserted-Identity: "Cullen Jennings" <sip:fluffy@cisco.com> P-Asserted-Identity: tel:+14085264000 Privacy: id When I use the following two (2) statements in a Freeswitch dialplan, the second "set," of course, overwrites the data stored by the first "set." <action application="set" data="sip_h_P-Asserted-ID=tel:+1404XXXXXXX"/> <action application="set" data="sip_h_P-Asserted-ID="George F Phelps" <sip:1404XXXXXXX@YY.YY.YYY.YYY>"/> Thanks, George

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Adding X-CID custom header for sipcapture integration
by Juan Antonio Ibañez Santorum 07 Jan '15

07 Jan '15

Hello, I am trying to add custom X-CID header to copy call id from a-leg to b-leg in order to make work call flow in sipcapture. I set channel variable sip_h_X-CID to add it to b-leg, I can see it is setted up but outgoing b-leg request does not contain the header. If I set up sip_h_B-CID it works but sip_h_X-CID not. Do you know why? Any other way to add the header? Regards

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Testing inbound calls from LOTS of origin points?
by Nathan Neulinger 07 Jan '15

07 Jan '15

Anyone know of any good way to test inbound calls from lots of different origins? Similar to what you do with things like 'global dns propagation checker'? I'd like to be able to periodically validate inbound calls from a bunch of different sources - ideally taking different paths inbound. (We've had a number of occasions lately where we suspect a routing problem external to our sip carrier only affecting certain inbound call sources.) I realize I can probably do this by hand, just by getting SIP trunks from a bunch of different providers, but is there any way that is a bit less manual/time consuming/third party to do the heavy lifting/etc? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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BLF status not always correct
by Raphael Lechner 07 Jan '15

07 Jan '15

Hi, Sometimes we have the issue that the status for the BLF linekeys don’t have the correct status. We have the same issue on different installations (different FreeSWITCH 1.4.X versions and different phone vendor/types, snom/yealink). Today we found the issue on a Installation with FreeSWITCH 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit). sip_subscriptions and sip_presence tables(at that moment the phone don’t have the right status) https://pastebin.freeswitch.org/23834 <https://pastebin.freeswitch.org/23834> Is that a known issue? How can I debug that and is there a workaround/solution for that? After rebooting the phones the status is working again and after some days the issue is back. On newer versions > 1.4.8 the problem is don’t occur so frequently. Thank you, Raphael

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lua custom variables
by David Villasmil Govea 07 Jan '15

07 Jan '15

Hello Guys, I have this lus script, which sets a custom variable, this is ok. The the call is transferred to another extension, and the variable is not there anymore, is this by design? Thanks! David -- DVG -- Imagination is more important than knowledge Albert Einstein

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Webrtc no video
by greg＠jazzmessengers.com 07 Jan '15

07 Jan '15

Hi folks, I have just installed freeswitch with a basic configuration. Actually, i trying to get to work a call with video. I'm using sipjs to set up my client. Each call has audio but no video. After some days checking the docs and some test, i don't really know where could come from this problem. Maybe the codec. Here, from fs_cli, the codecs list i have installed: === codec,ADPCM (IMA),mod_spandsp codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.722,mod_spandsp codec,G.723.1 6.3k,mod_g723_1 codec,G.726 16k,mod_spandsp codec,G.726 16k (AAL2),mod_spandsp codec,G.726 24k,mod_spandsp codec,G.726 24k (AAL2),mod_spandsp codec,G.726 32k,mod_spandsp codec,G.726 32k (AAL2),mod_spandsp codec,G.726 40k,mod_spandsp codec,G.726 40k (AAL2),mod_spandsp codec,G.729,mod_g729 codec,GSM,mod_spandsp codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,LPC-10,mod_spandsp codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,CORE_SPEEX_MODULE 27 total. ==== I tried to call to soft sip (x-lite on my computer, zoiper on my iphone 4s). here some log call accepted: ============== 2015-01-07 01:14:43.534992 [NOTICE] sofia.c:7475 Channel [sofia/internal/sip:1002@62.57.238.211:36178] has been answered 2015-01-07 01:14:43.534992 [DEBUG] switch_channel.c:3689 (sofia/internal/sip:1002@62.57.238.211:36178) Callstate Change RINGING -> ACTIVE 2015-01-07 01:14:43.554994 [DEBUG] switch_core_codec.c:246 sofia/internal/1000(a)37.187.113.94 Restore previous codec PCMA:8. 2015-01-07 01:14:43.554994 [DEBUG] mod_sofia.c:780 Local SDP sofia/internal/1000(a)37.187.113.94: v=0 o=FreeSWITCH 1420564044 1420564046 IN IP4 37.187.113.94 s=FreeSWITCH c=IN IP4 37.187.113.94 t=0 0 m=audio 25628 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv =============== when i active the video from one of the client: ==== 2015-01-07 01:18:23.094992 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1002@62.57.238.211:36178 [BREAK] 2015-01-07 01:18:23.094992 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1002@62.57.238.211:36178 [BREAK] 2015-01-07 01:18:23.114995 [DEBUG] sofia.c:6614 Channel sofia/internal/sip:1002@62.57.238.211:36178 entering state [received][100] 2015-01-07 01:18:23.114995 [DEBUG] sofia.c:6624 Remote SDP: v=0 o=- 13065063286487275 4 IN IP4 62.57.238.211 s=X-Lite release 4.7.1 stamp 74247 c=IN IP4 62.57.238.211 t=0 0 m=audio 55746 RTP/AVP 8 0 101 125 100 9 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:125 opus/48000/2 a=fmtp:125 useinbandfec=1 a=rtpmap:100 speex/16000 m=video 58218 RTP/AVP 115 34 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2 a=rtcp-fb:* nack pli 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3543 Set telephone-event payload to 101 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:2]/[PCMA:8:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [opus:125:48000:20:0:2]/[PCMU:0:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [speex:100:16000:20:0:1]/[PCMA:8:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [speex:100:16000:20:0:1]/[PCMU:0:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3881 Set 2833 dtmf send payload to 101 2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:5124 Audio params are unchanged for sofia/internal/sip:1002@62.57.238.211:36178. 2015-01-07 01:18:23.114995 [DEBUG] sofia.c:7259 Processing updated SDP 2015-01-07 01:18:23.114995 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1002@62.57.238.211:36178 [BREAK] 2015-01-07 01:18:23.134997 [DEBUG] sofia.c:6614 Channel sofia/internal/sip:1002@62.57.238.211:36178 entering state [completed][200] 2015-01-07 01:18:23.314994 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1002@62.57.238.211:36178 [BREAK] 2015-01-07 01:18:23.314994 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1002@62.57.238.211:36178 [BREAK] 2015-01-07 01:18:23.314994 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:1002@62.57.238.211:36178 [BREAK] 2015-01-07 01:18:23.334992 [DEBUG] sofia.c:6614 Channel sofia/internal/sip:1002@62.57.238.211:36178 entering state [ready][200] ==== Thank in advance for your help Greg

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getting call duration with event socket
by Bruce Lefko 07 Jan '15

07 Jan '15

Hi there. I'm trying to get the call duration in seconds using ESL, but it looks like the channel variables are not being set correctly. In CHANNEL_HANGUP events, I see the following variables: 'Caller-Profile-Created-Time' => '1420586926995761', 'Caller-Channel-Created-Time' => '1420586926995761', 'Caller-Channel-Answered-Time' => '1420586929503733', 'Caller-Channel-Progress-Time' => '0', 'Caller-Channel-Progress-Media-Time' => '1420586929203735', 'Caller-Channel-Hangup-Time' => '0', I would expect to generate the duration of the call by subtracting Caller-Channel-Answered-Time from Caller-Channel-Hangup-Time, but the hangup time is 0. The actual call in this case was 45 seconds long. What can I do use Freeswitch events to compute call duration? Thanks!

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IPV6 phone question about dial_string
by Mario G 06 Jan '15

06 Jan '15

I got calls to go to an IPV6 phone by copying this line from default.xml: <param name="dial-string" value="{sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> to the user extension.xml and changed it to: (this works fine) <param name="dial-string" value="{sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(internal-ipv6/ ${dialed_user}@${dialed_domain})}”/> my question is do I need to add internal-ipv6 also to presence_id like this: <param name="dial-string" value="{sip_invite_domain=${dialed_domain},presence_id=internal-ipv6/${dialed_user}@${dialed_domain}}${sofia_contact(internal-ipv6/ ${dialed_user}@${dialed_domain})}”/> Mario G

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