November 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Sofia gateway variables
by Stanislav Sinyagin 03 Nov '12

03 Nov '12

I've got the following SIP gateway which registers successfully at the ITSP: <gateway name="sipcall_41325133196"> <param name="username" value="41325133196"/> <param name="password" value="XXXXXXXXXX"/> <param name="extension" value="0325133196"/> <param name="expire-seconds" value="125"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="proxy" value="free2.voipgateway.org"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="false"/> <param name="ping" value="27"/> <variables> <variable name="sbc_callee_ccode" data="41" direction="inbound"/> <variable name="sbc_caller_ccode" data="41" direction="inbound"/> </variables> </gateway> Then the call from PSTN goes into the proper context, and the first condition in the context is: <extension name="call_debug" continue="true"> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> <action application="log" data="INFO Country code: ${sbc_callee_ccode}"/> </condition> </extension> Neither the info, nor the log show my two variables on inbound calls. Am I doing something wrong or is it a bug? I'm using git commit 94664868a8 in v1.2.stable branch thanks, stanislav

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freeswitch talk
by succer110＠tiscali.it 02 Nov '12

02 Nov '12

Hi! I'm a freeswitch user, and i have a little experience in developing freeswitch modules.Within a month, I'm goint to do a talk in a conference, focused on freeswitch. Specifically, i was thinking about this topics: -difference between FS and other pbx in the market-Where is the best place for FS in a VoIP architecture? (registrar, media encoding, application server, ivr, etc..)-how to scale If you have some *in depth* documentation about this topics i will be very glad.I've found a lot of blog post about comparison, but i didn't found a specific analysis of code and benchmarks between various app.(in this days i'll watch a just discovered source of information: ClueCon channel on youtube :D ) Also, if you want to suggest some new topics to talk about, it would be very cool for me to study something new, if it's a good way for helping the community :)thank you! Invita i tuoi amici e Tiscali ti premia! Il consiglio di un amico vale più di uno spot in TV. Per ogni nuovo abbonato 30 € di premio per te e per lui! Un amico al mese e parli e navighi sempre gratis: http://freelosophy.tiscali.it/

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I think its FreeForAll Time
by Ken Rice 02 Nov '12

02 Nov '12

Sure we shoot for 4PM Eastern.... But lets get it started now.... sip:888@conference.freeswitch.org as usual... Get in here... Today¹s Topic: TACOS! (or something) -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Just starting
by Rob Boutin 02 Nov '12

02 Nov '12

Hello, I am just starting out with FreeSWITCH and I am researching it for a project at work that will hopefully end up with full implementation of FreeSWITCH and had a few short questions as I am having difficulties getting it off the ground. I am currently running FreeSWITCH on a Windows 7 enterprise 64-bit system. My tiny test network of phones includes an xlite softphone on the windows machine 2 Linksys SPA922 phones and 1 Linksys SPA942 phone. They are being routed with a simple Linksys Etherfast router. My issue is that only one phone at a time will ever be registered and able to make calls. I have four extensions configured with identical settings 300 - 303. The peculiar thing is only the phone assigned to 301 will ever have a dial tone and be able to place calls. It doesn't matter which device I move 301 to it is only 301 that ever works. I was hoping to get some light shed on this, as it is likely related to some simple setting I over looked and would like to continue with my research. Thanks in advance, RJB

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Re: [Freeswitch-users] Passing dtmf to an IVR behind Google Voice problem
by Dmitry R 02 Nov '12

02 Nov '12

I've just tested the same configuration under asterisk 11 with chan_motif/xmpp. Everything works great. DTMF codes get thru to IVR behind a Google Voice entrance. This means, that either FS dingaling driver doesn't work correctly or there is a mess in dtmf signaling between dingaling and sofia. Is there any way to fix this? Dmitry _____ From: Dmitry R [mailto:rdmitry0911@gmail.com] Sent: Thursday, November 01, 2012 10:06 AM To: 'FreeSWITCH Users Help' Subject: RE: [Freeswitch-users] Passing dtmf to an IVR behind Google Voiceproblem No, that's not my case. I can't receive dtmf while that workaround is about sending dtmf codes Dmitry _____ From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, November 01, 2012 12:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Passing dtmf to an IVR behind Google Voiceproblem For the sake of posterity (and search engines) there's a workaround mentioned here: http://wiki.freeswitch.org/wiki/Dingaling#How_do_I_send_DTMF_with_Googletalk .3F -MC On Wed, Oct 31, 2012 at 11:29 AM, Dmitry R <rdmitry0911(a)gmail.com> wrote: Hi, does anybody know how to pass dtmf codes to an IVR behind the Google Voice entrance number? A simple example code from http://wiki.freeswitch.org/wiki/Google_Voice <extension name="google_in" continue="true"> <condition field="caller_id_name" expression="^(Google Voice)$" > <action application="log" data="Google Voice Call Incoming" /> <action application="set" data="execute_on_answer=send_dtmf 1"/> </condition> </extension> doesn't work for me. DTMF codes can't get thru. Everything else other then that works fine. Any help would be very much appreciated, Dmitry _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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mod_callcenter agents didn't change to Receiving
by Vbvbrj 02 Nov '12

02 Nov '12

Hello. Got a problem when using mod_callcenter. Agents where in list at state 'Available - Waiting'. They where tiered to a queue. Suddenly clients were queued but no agent were changed to Receiving. Mod_callcenter just kept clients in queue and didn't offer any agent to them. There were an agent for test which were logged out. When this agent logged in, it started to receive clients. The other agents remained as Waiting. When this test agent were logged out, again clients just sit in queue and agents with status Waiting were not tried to bridge with clients. Even a complete restart of FS didn't help. I had to delete these agents from agent's list completely with "callcenter_config agent del" and add them again. What happened, why I had to completely remove agents from agent's list and not use agent log out? In log didn't appear anything. Just info that to clients the position is announced. No info about trying any agent. -- Mimiko desu.

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survey after the call.
by Pankaj Belsare 02 Nov '12

02 Nov '12

Hi, i am trying out a dialplan to take survey and append the database with the responses and the call is hangup by the "agent" , can any body help with suggestion/example Pankaj

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mod_spandsp fails to load
by Nelson Camargo 01 Nov '12

01 Nov '12

I'm having issues to get mod_spandsp to work on one of my boxes, every time I try to load I get: 2012-11-01 14:48:19.199721 [CRIT] switch_loadable_module.c:1310 Error Loading module /usr/local/freeswitch/mod/mod_spandsp.so **/usr/local/freeswitch/mod/mod_spandsp.so: undefined symbol: jbg_enc_out** It's happening with both HEAD master and v1.2stable, any ideas? Thanks, Nelson

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FreeTDM with old a101
by Spencer Thomason 01 Nov '12

01 Nov '12

Hello, I have an old Sangoma a101 that I'd like to use with FreeSWITCH. When I try to configure it, I receive the error "Old a101/102 card not supported in this release!!! with FreeSwitch". If anyone can help out with any versioning (Wanpipe, Kernel, Distro, etc.) info that would work, I'd be greatly appreciative! It's really old I know :-) #wanrouter hwprobe ------------------------------- | Wanpipe Hardware Probe Info | ------------------------------- 1 . AFT-A101c : SLOT=3 : BUS=1 : IRQ=28 : CPU=A : PORT=PRI : V=25 Sangoma Card Count: A101-2=1 Thanks, Spencer

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