January 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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

3 2

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!

2 2

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

1 0

Google Voice Support?
by George F. Phelps 06 Jan '15

06 Jan '15

Now that Google has de-supported its XMPP connectivity to Google Voice, what is the Freeswitch configuration for interfacing to Google Voice? Thanks, George

5 9

Control of IPv6 vs. IPv4
by George F. Phelps 06 Jan '15

06 Jan '15

My VoIP provider supports both IPv4 and IPV6 registrations. I can only have one registration active at a time. Freeswitch is attempting both IPv4 and IPv6 connections. Randomly, sometimes a IPv4 connection is the first (only) registration established; other times it is the IPv6 connection. How to I configure Freeswitch to deterministically only attempt one type (my choice of either IPv4 or IPv6) of connection? freeswitch(a)ip-172-31-33-109.ec2.internal> sofia status Name Type Data State ============================================================================ ===================== external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external-ipv6::switch2voip.us gateway sip:XXXXXXXXXX@66.33.147.150 REGED external profile sip:mod_sofia@54.174.255.168:5080 RUNNING (0) external::switch2voip.us gateway sip:XXXXXXXXXX@66.33.147.150 TRYING (retry: 20s) 172.31.33.109 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@54.174.255.168:5060 RUNNING (0) ============================================================================ ===================== 4 profiles 1 alias freeswitch(a)ip-172-31-33-109.ec2.internal> Thanks, George

3 8

Detecting multiple tones with freeswitch
by Bunea Lucian 06 Jan '15

06 Jan '15

Hello, My SIP provider (Vodafone) decided to make me a Christmas present: they change the configuration of their SIP server. Using the old configuration they were sending SIP codes for decline, busy and unavailable. Now they are sending early media with tones (for decline and busy) and a one minute message for unavailable. Each status is followed by a SIP/480. Since I need to be able to distinguish between different statuses, I have determined the following: - if I don't receive a ring tone within 7 second, the called number is unavailable; - if I receive a busy tone within 4 seconds the called number is busy; - if I receive a busy tone after 4 second the call number has declined the call; This is how I tried to implement it: - after receiving early media (SIP/180 or SIP/183) a timer is started for call hangup (sched_hangup) - if I receive a ring tone within 7 seconds the timer is canceled (sched_cancel); - if I receive a busy tone a lua script is called; <action application="set" data="sip_ignore_remote_cause=true"/> <action application="set" data="execute_on_media=sched_hangup +7 SUBSCRIBER_ABSENT"/> <action application="tone_detect" data="busy 425 w +60000 lua 'busy_detected.lua ${uuid} ${created_time}' 2"/> <action application="tone_detect" data="ring 450 w +7000 sched_cancel"/> <action application="bridge" data="sofia/external/${destination_number}(a)XXX.XXX.XXX.XXX"/> But, for some reason, the tones are detected, but the applications are never called. If I add one of the following lines to the dialplan the corresponding application is called: <action application="set" data="execute_on_tone_detect=sched_cancel"/> or <action application="set" data="execute_on_tone_detect=lua busy_detected.lua ${uuid} ${created_time}"/> but I can't tell which tone was detected... What am I missing? PS: Tested with FreeSWITCH Version 1.4.14-1~64bit. Regards, Lucian

3 4

FreeSWITCH Week in Review (Master Branch) December 28th-January 4th
by Ken Rice 06 Jan '15

06 Jan '15

New Post on freeswitch.org from kathleen check it out at http://ift.tt/17g3uGI FreeSWITCH Week in Review (Master Branch) December 28th-January 4th Hello, again. This week in the FreeSWITCH master branch we had 10 commits. The features for this week are improvements to socket handling and the implementation of grunt for javascript management in mod_verto. Grunt is a task runner for javascript designed to automate small mundane tasks like minification, compilation, unit testing, and linting. New features that were added: bf5210b Improved mod_verto socket handling code 0db3b1e FS-7127 Add grunt to mod_verto [Jira: http://ift.tt/17g3xlC] Improvements to documentation: 7a517ee FS-7127 Update README and sync re-generate verto-min [Jira: http://ift.tt/17g3xlC] f0ec193 FS-7127 Update the README for mod_verto [Jira: http://ift.tt/17g3xlC] The following bugs were squashed: 65631ed Revert FS-7004 pending updated fixed for that issue from the original author [Jira: http://ift.tt/1pgbVcN] a067a49 FS-7046 fix warning introduced from b341ff7 properly [Jira: http://ift.tt/1yJc4rg] a961b0e Fix return size in mod_verto 1965b3b FS-7106 #resolve Fix concurrency issue in mod_httapi [Jira: http://ift.tt/1BEamXd] 0bec209 Fix fsapi in verto

1 0

Canceling sched_hangup and UUID lookup
by Kurtis Heimerl 06 Jan '15

06 Jan '15

Hello Freeswitch-users, I'm building a group billing solution for our freeswitch system. Our account management is external, so I need to make something custom. I have two questions about how to do some stuff in FS. 1) I know how to limit call length (sched_hangup). I can query an external account via REST to see how many credits they have and limit the call with that. However, when a second user on the same account calls in, I need to change that timeout. Is there a way to cancel a sched_hangup? I found sched_cancel, but that seems to be for broadcasts and transfers, neither of which I'm doing. I *guess* I could transfer to a hangup extension instead of hanging up... but that seems awful ugly. 2) Is there a way to "tag" a call (uuid) with an arbitrary text string and then look it up by that? That would allow me to tag each call with the account name and later get all calls under that account. Thanks!

4 10

How to play a stream other than mp3 with mod_shout.
by Aqs Younas 06 Jan '15

06 Jan '15

Hi, All How can i play a live stream other than mp3 with mod_shout or any module.? Is there any way to buffer the stream before playing it with mod_shout. Currently i have a list of streams and when i play them with mod_shout some work fine but others give (time out) error. How can i play mostly stream in freeswitch? Thanks

8 16