April 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Come see our own Ken Rice on the voice users conference today!
by Anthony Minessale 19 Apr '13

19 Apr '13

http://vuc.me #vuc on freenode Starts in 30 min. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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FreeSWITCH + Raspberry Pi = SwitchPi today on the VUC Conference
by Ken Rice 19 Apr '13

19 Apr '13

Hey guys, time for a little self promotion. I will be speaking on the VUC Conference today, Topic will be running FreeSWITCH on the Raspberry PI for use as a hard phone. There will be some technical discussion as well as showing off the hardware. Join us at 12:00 Noon Eastern via http://vuc.me/live I suggest using the link to watch via YouTube. Also the 888 conference bridge will be linked for those wanting to listen. (sip:888@conference.freeswitch.org) -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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fusionpbx not staying logged in
by Jason Holden 19 Apr '13

19 Apr '13

I dug through the Wiki and hoping this isn't too off topic but does anyone know why with fusionpbx once installed and I log in it keeps logging out asking for the username and password once I click on an item?

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mod_fifo ongoing calls information
by Ashish gautam 19 Apr '13

19 Apr '13

Hi All, I want to get information about which members are talking to which caller in real time. How can I do that? Please throw some light. Thanks.

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State of Call Progress Analysis (CPA)
by Lynn Nielson 19 Apr '13

19 Apr '13

I am looking for comments and opinions on the current state of Call Progress Analysis (CPA) with FreeSwitch. I believe Sangoma has a solution with their Lyra product but that a current module for that product is still in development. At one time, work was being done with LumenVox, but that has stopped. There was a module (mod_woomera), part of Dialogic, but it is over two years old. And there is the VMD and AVMD modules that indicate a high processing load to run effectively. Mostly, it seems that any CPA used needs to run separate from FreeSwitch in a proxy or re-directed mode to be effective under any kind of heavy load. A few years ago, this topic was popular, but lately it has either been regulated out of existence or found to be in-effective. Is there anyone with experience using call progress analysis or opinions on the topic? Thanks, -- Lynn Nielson Green Seed Technologies

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G729 codec INCOMPATIBLE_DESTINATION
by Jimmy Chang 19 Apr '13

19 Apr '13

Hi, I used FreeSWITCH Version 1.3.17+git~20130312T033518Z~7d29a92f55 and client could talk with G729 without problem. Today I install a new FS and the version is FreeSWITCH Version 1.5.1b+git~20130417T010754Z~538d9beacd. I copied the entire conf directory in old FS to new FS. I got the problem. 2013-04-18 21:25:06.899049 [DEBUG] sofia.c:5694 Remote SDP: v=0 o=Dialogic_SIP_CCLIB 62147888 62147889 IN IP4 10.0.110.41 s=Dialogic_SIP_CCLIB c=IN IP4 10.0.110.41 t=0 0 m=audio 49152 RTP/AVP 18 101 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 2013-04-18 21:25:06.899049 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G729:18:8000:30:8000]/[G729:18:8000:20:8000] 2013-04-18 21:25:06.899049 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [telephone-event:101:8000:30:0]/[G729:18:8000:20:8000] 2013-04-18 21:25:06.899049 [DEBUG] switch_core_media.c:2671 Substituting codec G729@30i@8000h 2013-04-18 21:25:06.899049 [DEBUG] switch_core_media.c:1617 No audio codec available 2013-04-18 21:25:06.899049 [DEBUG] switch_core_media.c:2752 Set 2833 dtmf send payload to 101 2013-04-18 21:25:06.899049 [NOTICE] sofia.c:5812 Hangup sofia/ivvr/9999110 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] Any advice? Thanks. Jimmy

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uuid_bridge causes DESTINATION_OUT_OF_ORDER with FS rev 1.2.7
by Siri MM 19 Apr '13

19 Apr '13

Hello All, I am working on migrating our call system from FreeSWITCH 1.0.6 to 1.2.7. The setup is as follows: * CallerDevice registered with Remote FreeSWITCH * CalleeDevice registered with Local FreeSWITCH * Local FreeSWITCH and Remote FreeSWITCH are in different subnets I have a C application which is connected to, and interacts with FreeSWITCH. On receiving a call from Remote FreeSWITCH on Local FreeSWITCH, the call is parked, and then a couple of Lua scripts are executed: 1. first one sends an INVITE to the callee device via freeswitch interface, which makes the callee device to ring 2. Once the call is accepted on the device, a second script executes, which calls the uuid_bridge applicaiton to bridge the parked call UUID and the callee UUID However, most of the times (it does work a few times), the call establishment fails with a DESTINATION_OUT_OF_ORDER error. This exact setup, when run with the 1.0.6 version, works without any glitches. When I peeped through the code, I see that, the call state of the callee leg, after going into HIBERNATE STATE, never recovers, because of which the call times out in the method uuid_bridge_on_soft_execute(), and throws this error. In case of the 1.0.6 version, this doesn't happen, and 'State Change CS_HIBERNATE -> CS_RESET' takes place, and the call continues. The uuid_bridge_on_hibernate() handler, for some reason, doesn't seem to be getting called. Would appreciate any inputs on how to resolve this.. I probably need to update my application to work with 1.2.7 version, but am not sure! Logs at : http://pastebin.freeswitch.org/20800 Thanks!

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Integration with Vocality
by nordin mohamed 19 Apr '13

19 Apr '13

Hi all, Actually I'm quite new with Freeswitch. Currently I'm trying to create a Comm Server that able to provide a conference to voip phone and radio. I'm using Basics Radio Relay(Vocality) as the radio repeater. Basics Radio Relay is using the silent suppression frame as marker to keying the PTT to the radio. Asterisk currently not able to support silent suppression. I thought of using Freeswitch as I know it support for silent suppression. Is there anybody have experience of integration freeswitch and Basic Radio Relay. thank you -nordin mohamed

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Do you use Adhearsion? If so, how?
by Ben Langfeld 19 Apr '13

19 Apr '13

Adhearsion got FreeSWITCH support last year, and it's been awesome since; we've been doing a lot more work with FreeSWITCH and several people have switched projects from Asterisk to FreeSWITCH as a result. Now we'd like to ask a favour of anyone who has used Adhearsion from the FreeSWITCH community. We'd like your feedback in order to help set our course for the future. Only 10 minutes of your time could help the project immeasurably, so please do complete our survey about usage patterns here: https://www.surveymonkey.com/s/53RGSY8 Thanks in advance, Ben Langfeld Adhearsion Foundation

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OPUS in FreeSWITCH
by thomas peterseil 18 Apr '13

18 Apr '13

hello, i just installed the actual version of freeswitch and now i wanted to make opus working. i added and activated the mod_opus module and now i wanted to ask, what else i have to do to make opus work with my freeswitch? i tried to add opus in the vars.xml file, but it didn´t help. what´s the exact syntax i have to use to enable opus in the vars.xml file? are there some other changes i have to do to make it working? thank you very much for your help! best, thomas

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