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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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mod_limit does not reset count when call gets to voicemail?
by Yehavi Bourvine 15 Oct '10

15 Oct '10

Hello, We are using mod_limit in order to track user's state and know when to play busy/call-wait tone. We encountered two problems (which are probably related): - When the call gets to the voicemail the extension is still though as busy until the caller hangs. - If a call is picked-up using the intercept application the original destination is still thought as busy. When looking in the limit_data table the record of the call is indeed there until a final call hangup. According to the wiki we tried setting the variable limit_ignore_transfer to false both with set command before calling the limit application and also in limit.conf.xml. However, this does not change this behaviour. Is this the intended behaviour? Can it work as I want? (i.e. once the call arrived to voicemail or picked-up the original destination is marked as free). We are using 1.0.6; since this is a production system it is quite hard for me to test the latest git. Thanks! __Yehavi:

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Java_ESL_Client - unhandled exception caught
by Vedran Zeljeznak 15 Oct '10

15 Oct '10

hello, i've found a bug in Java_ESL_Client and the info on the website (http://wiki.freeswitch.org/wiki/Java_ESL_Client#Issues) said it should be reported here so i'm sending it. It looks like sometimes FS sends malformed URL encoded values through Event Socket, probably because value is not properly terminated (probably on my version of FS only - FreeSWITCH Version 1.0.6 (hacked-20100420T130612Z) - official version). exception trace follows: #### TRACE / LOG info 14.10.2010. 11:16:29 org.jboss.netty.channel.SimpleChannelUpstreamHandler WARNING: EXCEPTION, please implement org.freeswitch.esl.client.inbound.InboundClientHandler.exceptionCaught() for proper handling. java.lang.IllegalArgumentException: URLDecoder: Illegal hex characters in escape (%) pattern - For input string: "\5" at java.net.URLDecoder.decode(URLDecoder.java:173) at org.freeswitch.esl.client.transport.event.EslEvent.parsePlainBody(EslEvent.java:187) at org.freeswitch.esl.client.transport.event.EslEvent.<init>(EslEvent.java:73) at org.freeswitch.esl.client.transport.event.EslEvent.<init>(EslEvent.java:62) at org.freeswitch.esl.client.internal.AbstractEslClientHandler.messageReceived(AbstractEslClientHandler.java:80) at org.freeswitch.esl.client.internal.debug.ChannelEventRunnable.run(ChannelEventRunnable.java:76) at org.jboss.netty.handler.execution.MemoryAwareThreadPoolExecutor$MemoryAwareRunnable.run(MemoryAwareThreadPoolExecutor.java:541) at java.util.concurrent.ThreadPoolExecutor$Worker.runTask(ThreadPoolExecutor.java:886) at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:908) at java.lang.Thread.run(Thread.java:619) #### TRACE / LOG info --- Vedran Zeljeznak

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tls/ssl gateway insists on registering SIP
by Jerre Cope 15 Oct '10

15 Oct '10

I'm attempting to implement the Hybrid model as shown on http://wiki.freeswitch.org/wiki/Tls#Hybrid_Encryption FS(Truck)=ssl/tls=>FS(Office) My idea was to create a dial plan so that when FS(Truck) dials 999, a secure connection could be made to the office. The pastebin shows my directory entry in FS(Office) and the external gateway in FS(truck) http://pastebin.freeswitch.org/14245 The gateway insists on connecting on 5060, calls made through the gateway appear in tshark to be using tls. I can block 5060 at the FS(office) firewall after registration and make calls through the gateway OK until FS(truck) tries to register again. The FS(truck) default gateway is voip.ms, the idea being non-secured calls would go out the voip.ms gateway so as to limit the traffic to FS(office). I'm operating way beyond my experience, so my attempt at this may be a goofy start. I'm asking for a little direction/perspective on how to implement. I'm using ssl because all my FS(trucks) will be dynamic IPs. Thanks for all the really cool programming/documentation so far. I'm having great fun!

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