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Ring timer
by Andre 01 Nov '13

01 Nov '13

Is the call_timeout a timeout for ring timeout? I see this note: WARNING: Beware that if you are not using {ignore_early_media=true} call_timeout is no longer applicable as soon as early media signal is received. I am not setting Early Media. What timeout should I use to let the call ring for x seconds than hang up? Thanks

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mod_lcr and UnixODBC Error
by FERNANDO VILLARROEL 31 Oct '13

31 Oct '13

Dear all. I am trying to use mod_lcr, but in my tests i did received the follow error: 2013-10-31 16:35:03.916887 [ERR] switch_core_sqldb.c:1114 ERR: [SELECT l.digits AS lcr_digits, c.carrier_name AS lcr_carrier_name, l.rate AS lcr_rate_field, cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix, l.lead_strip AS lcr_lead_strip, l.trail_strip AS lcr_trail_strip, l.prefix AS lcr_prefix, l.suffix AS lcr_suffix, cg.codec AS lcr_codec, l.cid AS lcr_cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND ( (digits IN (12145551111, 1214555111, 121455511, 12145551, 1214555, 121455, 12145, 1214, 121, 12, 1) AND lrn = false) OR (digits IN (12145551111, 1214555111, 121455511, 12145551, 1214555, 121455, 12145, 1214, 121, 12, 1) AND lrn = true)) AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC, rate, quality DESC, reliability DESC, random();] [STATE: 42883 CODE 7 ERROR: [unixODBC]ERROR: el operador no existe: prefix_range = bigint; Error while executing the query ] I am using PostgreSQL 8.4.11 and Debian Squezee How i can solve this issue? Regards

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bind_digit_action stops working when the call is sent to a socket
by Ali Pey 31 Oct '13

31 Oct '13

Hello, As soon as the call is sent to a socket, bind digit actions stop working. <action application="socket" data="127.0.0.1:9090 full"/> I have tried setting bind_digit_action both before and after the call is sent to the socket. It doesn't work. Before call is sent to socket, I see: 2013-10-31 10:48:38.183166 [DEBUG] switch_rtp.c:3915 RTP RECV DTMF 5:3520 2013-10-31 10:48:38.183166 [DEBUG] mod_dptools.c:186 sofia/sip-trunks/ 6132952508(a)qasip.onebox.com Digit match binding [exec:playback][ivr/ivr-please_enter_extension_followed_by_pound.wav] After is sent to the socket, it becomes: 2013-10-31 11:07:38.042566 [DEBUG] switch_rtp.c:3915 RTP RECV DTMF 5:4160 2013-10-31 11:07:38.042566 [DEBUG] switch_channel.c:419 RECV DTMF 5:4160 2013-10-31 11:07:38.042566 [DEBUG] mod_dptools.c:2049 Digit 5 Any ideas? Is this a bug? Is there a work around? Thanks, Ali Pey

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max-sessions value small may cause crash
by Ashou Han 31 Oct '13

31 Oct '13

Hi, My FS is FreeSWITCH Version 1.2.12+git~20130910T181609Z~f5ba0bb7d4 (git f5ba0bb 2013-09-10 18:16:09Z) I set <param name="max-sessions" value="20"/>, <param name="sessions-per-second" value="10"/> in switch.conf.xml FS carshed very offen like this in messages log: Oct 26 19:10:21 fs kernel: [4592903.781042] freeswitch[29353]: segfault at 0 ip 00007f4e22e0c8cd sp 00007f4e0924e5b0 error 4 in libfreeswitch.so.1.0.0[7f4e22ce4000+229000] Oct 26 23:41:45 fs kernel: [4609190.441226] freeswitch[518]: segfault at 0 ip 00007ff71baf58cd sp 00007ff7068ae5b0 error 4 in libfreeswitch.so.1.0.0[7ff71b9cd000+229000] Oct 28 09:57:58 fs kernel: [89606.041406] freeswitch[23780]: segfault at 0 ip 00007f68651d78cd sp 00007f684ff205b0 error 4 in libfreeswitch.so.1.0.0[7f68650af000+229000] Oct 29 16:55:24 fs kernel: [201076.501294] freeswitch[16422]: segfault at 0 ip 00007f0b68fc48cd sp 00007f0b4f50d5b0 error 4 in libfreeswitch.so.1.0.0[7f0b68e9c000+229000] lots in freeswitch.log: 2013-10-25 15:25:24.651750 [CRIT] switch_core_session.c:2228 Over Session Limit! 20 2013-10-25 15:25:24.651750 [CRIT] mod_sofia.c:4869 Error Creating Session 2013-10-25 18:36:03.311729 [CRIT] switch_core_session.c:2228 Over Session Limit! 20 2013-10-25 18:36:03.311729 [CRIT] mod_sofia.c:4869 Error Creating Session ... 2013-10-30 15:01:46.501223 [CRIT] mod_sofia.c:4869 Error Creating Session 2013-10-30 15:01:46.521224 [CRIT] switch_time.c:1007 Over Session Rate of 10! May this cause the FS crash? I have another FS server with same Hardware and FS version, but <param name="max-sessions" value="2000"/> and no crash issues. Thanks, Ashou Han

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Code negotiation based on local priority
by Samir Doshi 31 Oct '13

31 Oct '13

Hi Guys, I have question about code negotiation priority. Here is the case, In fs server we have set local codec priority g729, g711, g723, but client will send us call in g711, g723, g729 priority. so here fs will negotiation g711 as that's client codec priority. But can i be able to handle priority based on my local fs server? ----- Thanks, Samir -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Code-negotiation-based-on-loc… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Video only call
by Limit 31 Oct '13

31 Oct '13

Good day! I'am experimenting with SIP video calls (using VP8 video codec). Ordinary audio+video calls works very well (audio recordered and video passed through). But if I try to make only video call, i.e. SDP contains only video media, than I got an error and channels hang up: 2013-10-31 17:00:38.494743 [DEBUG] sofia.c:5708 Channel sofia/internal/sip:1008@10.10.104.125:10018 entering state [ready][200] 2013-10-31 17:00:38.494743 [DEBUG] sofia_glue.c:5382 Video Codec Compare [VP8:99]/[H264:97] 2013-10-31 17:00:38.494743 [DEBUG] sofia_glue.c:5382 Video Codec Compare [VP8:99]/[VP8:99] 2013-10-31 17:00:38.494743 [ERR] sofia_glue.c:3025 No audio codec available 2013-10-31 17:00:38.494743 [DEBUG] sofia_glue.c:2975 Set VIDEO Codec sofia/internal/sip:1008@10.10.104.125:10018 VP8/90000 0 ms 2013-10-31 17:00:38.514745 [ERR] sofia.c:6464 RTP Error! 2013-10-31 17:00:38.514745 [NOTICE] sofia.c:6465 Hangup sofia/internal/sip:1008@10.10.104.125:10018 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER] 2013-10-31 17:00:38.514745 [DEBUG] switch_channel.c:3130 Send signal sofia/internal/sip:1008@10.10.104.125:10018 [KILL] Incoming SDP information looks like this: v=0 o=- 1383224434 1 IN IP4 10.10.104.125 s=Test/2.0beta263 c=IN IP4 10.10.104.125 t=0 0 m=video 5000 RTP/AVP 31 34 110 109 102 103 96 108 107 b=AS:768 b=AS:768000 a=rtpmap:31 h261/90000 a=fmtp:31 CIF=1;QCIF=1 a=rtpmap:34 H263/90000 a=fmtp:34 F=1;CIF=1;CIF16=1;CIF4=1;maxbr=7680;QCIF=1;SQCIF=1 a=rtpmap:110 H263-1998/90000 a=fmtp:110 D=1;F=1;I=1;J=1;CIF=1;CIF16=1;CIF4=1;maxbr=7680;QCIF=1;SQCIF=1 a=rtpmap:109 H264/90000 a=fmtp:109 max-fs=9000;max-mbps=270000;profile-level-id=428028 a=rtpmap:102 H264/90000 a=fmtp:102 packetization-mode=1;max-fs=9000;max-mbps=270000;profile-level-id=428028 a=rtpmap:103 H264/90000 a=fmtp:103 packetization-mode=1;max-fs=9000;max-mbps=270000;profile-level-id=640028 a=rtpmap:96 raw/90000 a=fmtp:96 rate=90000;height=288;width=352;colorimetry=BT601-5;depth=8;sampling=YCbCr-4:2:0 a=rtpmap:108 X-MX-VP8/90000 a=fmtp:108 x-mx-max-size=1920x1200 a=rtpmap:107 VP8/90000 a=fmtp:107 max-fs=9000 a=rtcp-fb:* pli fir tmmbr tstr freeswitch local SDP: v=0 o=FreeSWITCH 1383203862 1383203863 IN IP4 10.10.104.125 s=FreeSWITCH c=IN IP4 10.10.104.125 t=0 0 m=video 32038 RTP/AVP 98 99 a=rtpmap:98 H264/90000 a=fmtp:98 packetization-mode=1;max-fs=9000;max-mbps=270000;profile-level-id=640028 a=rtpmap:99 VP8/90000 a=fmtp:99 max-fs=9000 and remote SDP: v=0 o=- 1383224434 1 IN IP4 10.10.104.125 s=SimpleOPAL/3.12.7 c=IN IP4 10.10.104.125 t=0 0 m=video 5002 RTP/AVP 99 b=AS:16000 b=AS:16000000 a=rtpmap:99 VP8/90000 a=fmtp:99 max-fs=9000 a=rtcp-fb:* pli fir tmmbr tstr -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Video-only-call-tp7595976.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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core dump problems fs 1.2.14 oct 17
by Jason Holden 30 Oct '13

30 Oct '13

freeswitch v1.2.14 date october 17 Currently we are experiencing random crashes, can anyone provide advice why? I have a core dump I can send, not sure if the mail list will permit this file size?

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mod_cdr_mongodb: leg_a doesn't exist when calling between 2 local extensions neither on incoming calls on a tenant
by Nuno Reis 30 Oct '13

30 Oct '13

Hi guys. I'm changing my cdr database to mongo and i'm using mod_cdr_mongodb<http://fisheye.freeswitch.org/browse/%7Ebr=v1.4.beta/freeswitch.git/src/mod…> . Everything looked promising although i'm facing an issue: - When a call is placed between 2 local extension (on the same tenant), there is no leg_a on mongo's cdr table. - I also have the issue on incoming call (Ex: from a SIP gateway) I can only see everything as expected when placing outgoing calls (let's say on a SIP gateway). Can someone confirm this behavior. I'm using FS 1.2.12. Looking forward to hear from you. BR, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…>| www.wavecom.pt** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>

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UUID from Extension Number
by Jeremy Childs 30 Oct '13

30 Oct '13

I'm trying to create a "works-anywhere" eavesdrop extension. I've got the demo application that uses the hash/spymap functionality working correctly, but you would need the caller-id of the incoming call in order to eavesdrop on incoming calls. This could be fixed by capturing the destination number instead of the caller-id number for incoming calls (and I've done this, it works fine). However, this breaks in the case where an attended transfer has happened (the UUID of the call that gets connected "belonged" to the receptionist). The most robust thing I can think of is to do a live lookup of the call UUID using the "final" extension number. Is there any way to do this lookup from the dialplan? To be clear, what I'd like to do is this: Incoming call is picked up by receptionist (ext 100). Receptionist transfers call to extension 101. The Boss (extension 200), dials *88101 which looks up the UUID of extension 101's current call, provides this UUID to the eavesdrop extension, and the eavesdrop proceeds. There doesn't appear to be any way to do this from the dialplan, however -- am I missing something?

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Re: [Freeswitch-users] Raspberry Pi, mod_gsmopen GSM dongle
by Sergey Zhuravlov 30 Oct '13

30 Oct '13

I am glad to tell you that: 1) mod_gsmopen and e1550 work on Raspberry Pi (incoming, outgoing SMS in both directions). 2) and the stable version FS and the latest FS compile on Raspberry Pi, but it's a long time. 3) The latest version does not run on my raspberries. Need advice on optimizing the FS for raspberry Pi, as low-productivity device. Can any startup options? Maybe someone knows what modules you can disable resource-intensive. Something else? Does somebody have a positive experience with Raspberry Pi, mod_gsmopen > GSM dongle ??? > I use Raspberry Pi B and e1550. > > It's a shame that with an asterisk all works fine -- RasPBX. http://www. > raspberry-asterisk.org/ > > But I prefer the FreeSWITCH ;-) > On the basis of experience, I thought that PS will work best for this > modest hardware faster and require fewer resources. > > But it turns out it is not. Asterisk with a web server, MySQL database, > and other, less load than FS. > In this connection, other parameters may be used starting FS to save > resources? > > And the main question! Silence an incoming call on 5000 (IVR) as well as an > outgoing in gsm network. Those are not sound. > > -- > WBR, Sergey > > GTALK/JABBER:4orbit@gmail.com > > -- WBR, Sergey GTALK/JABBER:4orbit@gmail.com

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