October 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch Video Capture and Playback
by Esben Stien 18 Oct '10

18 Oct '10

I'm using ekiga with mod_fsv, trying to record and play back video. When I dial the record extension, it seems to record something, as the video file gets bigger. Trying then to dial the extension for play back, just hangs up, with freeswitch saying: od_fsv.c:247 File version does not match! There seems to be no information on the FSV format or the mod_fsv module on the wiki. Is this at all supposed to work?. What clients and codecs were successful?. Any pointers as to what I can try?. -- Esben Stien is b0ef@e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n

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mod_erlang_event core dump on load with shortname=false on Mac
by Seven Du 18 Oct '10

18 Oct '10

Hi, I experience this a few days ago, everything is ok with shortname=true, but shortname=false causes immediately core dump on load. I also pulled the latest git. problem remains. It has nothing to do with the latest change. The last change does help. I set <param name="nodename" value="freeswitch(a)192.168.1.128"/> then no more crash. So, the problem might be because it has trouble on get the fully qualified domain name on my box. Any other experience on this? I can make a jira if confirmed a bug. Thanks. -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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some questions of freeswitch & X-Lite
by 江 Chiang 至中 Chih-Chung Wybie 18 Oct '10

18 Oct '10

Hello I set two SIP servers 60.248.175.37 & 60.248.175.38 and use X-Lite to call my question is how can I call between these servers? if I can , how shoould I set in *.xml? or X-Lite? thank you for your attension Best Regards Gary

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How to change the language in conference
by Eli Hayun 18 Oct '10

18 Oct '10

Hi 1) I am trying to change the default language in the conference but no success I did : set default_language=ru but I still hear everything in English 2) How to change the language when I am running api:execute("conference"....) in Lua? Thanks -- Eli Hayun Hebrew University Jerusalem

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Compile on Windows
by Tim B 17 Oct '10

17 Oct '10

Anybody compiling freeswitch on windows? I am running windows 7 x64 with VS2010. Downloaded the latest git snapshot of freeswitch 1.0.6 and opened the vs2010 project, then tried to compile... getting like 700+ errors. I was able to compile version 1.0.4 on Vista with VS2008 with no problems awhile back. Just wondering if anyone else had any problems similiar and how they resolved them. Tim

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PHONO ...turns any web browser into a PHONE and IM Client
by EdPimentl 17 Oct '10

17 Oct '10

http://phono.com/ Phono is a simple JQuery plugin and JavaScript SDK that turns any web browser into a multi-channel communications platform capable of making phone calls and sending IM messages. You can even connect to SIP clients; all with a simple unified API. -E

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Read only filesystem
by Troy Anderson 17 Oct '10

17 Oct '10

I certainly don't know if this is a freeswitch specific issue, but we have several systems in the field running freeswitch on top of CentOS 5.5 and others on Ubuntu 10.04. Some of the systems, after a weeks of performance, go into a read only state on the hard drive, at which point, a restart is the only fix. After the reboot, the drive seems OK - I've run smartctl to verify that all seems good. Sometimes, in readonly mode, it even gives an I/O error when trying to execute the /sbin/shutdown and we have to manually reboot. At this point, I am trying to figure out what may be the root cause. Has anyone else experienced something this? Any suggestions? Thanks, Troy

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Passing incoming Caller ID
by Dave Stevenson 16 Oct '10

16 Oct '10

Hi, can someone help me with passing caller ID to FreeSwitch extensions please? I have a Linksys SPA-3102 setup to receive PSTN calls and Caller ID has just been configured on the line. The Gateway is setup as Extension 1000 with Dialplan entries <variable name="effective_caller_id_name" value="PSTN Line"/> <variable name="effective_caller_id_number" value="1000"/> I have a group name (primrose) setup (in directory\default.xml) to ring a number of phones I have extension 100 configured with <variable name="callgroup" value="primrose"/> The Gateway is setup to dial extension 100 on an incoming call When a call comes in, the Gateway sees the Caller ID, e.g, 07751234567 And FreeSwitch see the caller ID from the gateway :- [NOTICE] switch_channel.c:669 New Channel sofia/internal/07751234567(a)192.168.1.181 [f863c03f-0f7e-4e4b-beb0-dfa57dad2b50] When the group extension ring, the caller is shown as "PSTN Line" on the extensions. Can someone tell me how & where I can modify the config to pick up the CID from the Gateway and pass it to the group extensions please ? regards Dave

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iLBC codec generating only noise.
by xuyan yang 15 Oct '10

15 Oct '10

Hi, I am trying to use iLBC codec with both eyebeam and some iphone client. When a ivr is called, the client can here system voice and make dtmf input. but the voice recorded from client's microphone is only noise. The call between 2 clients also have such problem. Sometimes, FS may even got crashed with the following information: 2010-10-08 19:56:21.343750 [INFO] switch_rtp.c:2527 Auto Changing port from 10.20.132.244:18570 to 8 2.132.139.197:19536 alloc: asked for negative size -2147483648 in var.xml, I have added iLBC instead of iLBC@30i, because iLBC@30i is ignored during codec negotiation. <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h ,G722,speex@8000h@20i,iLBC,PCMU,PCMA,GSM"/> The git version last week is used in my test. Is there anything wrong with my setup? Thanks.

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UI Developer Job for CudaTel
by Anthony Minessale 15 Oct '10

15 Oct '10

We're expanding our horizons again over at CudaTel and hiring some more developers. Anyone interested, who lives in WI, OK, CA or MI or willing to move to one of those should apply. Looking for people with good HTML Design instincts and JavaScript / Jquery skills. Full-time job with challenging/rewarding work. Send inquires to jobs(a)freeswitch.org with a resume and any example work. -- 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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