October 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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IVR not available with iLBC
by Vincent Xia 30 Oct '13

30 Oct '13

i have configured FS to use iLBC for audio CODEC, it works perfect for any ordinary calls, but when making calls to IVR, the calls hangs up immediately after being established, the log on FS consloe is: [WARNING] mod_sofia.c:1279 Asynchronous PTIME not supported, changing our end from 20 to 30 [WARNING] switch_core_codec.c:704 Codec iLBC Exists but not at the desired implementation. 8000hz 30ms [ERR] sofia_glue.c:3062 Can't load codec? is it the problem that iLBC could not decode and play IVR sound files? so should i convert the .wav IVR sound files to iLBC format, or is there any better solutions since i may have to maintain two sets of sound files, one for iLBC and the other for other codecs as PCMU?

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FS 1.2.x mod_cdr_csv change creation mode
by Karsten Horsmann 30 Oct '13

30 Oct '13

Hello, the default creation mode of cdr csv files are 0600. Is there an configuration parameter to change that creation mode to 0640? -- Kind Regards *Karsten Horsmann*

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Prevent parking on top of another call
by Spencer Thomason 30 Oct '13

30 Oct '13

Hello, Does anyone have any suggestions as to prevent a blind transfer from parking on top of another call when using valet parking? Thanks! Spencer

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ClueCon Weekly News and Notes
by Ken Rice 29 Oct '13

29 Oct '13

Hey Guys, We had a great ClueCon Weekly Last week with Cal Leeming. This week we will be following up on last weeks Information with new information! Join us Wed at 1PM Eastern, 10AM Pacific via sip:888@conference.freeswitch.org or for other access see http://fs0.us/call888 And for all your social networking needs don¹t forget to join our circles on G+, our Group on FaceBook, and Follow us on Twitter! G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE Have a Great Week! Ken -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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