August 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Scheduler in module
by mark morreny 18 Aug '09

18 Aug '09

Hi, I would like to collect some data in my module which would listen to the event socket and then a scheduler would kick off every so often to clean up the data. Does anyone know what is the best way to implement a scheduler in a mod? Best Regards, Mark

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how to use freeswitch as a sip client
by Tzury Bar Yochay 18 Aug '09

18 Aug '09

Hi, Can someone instruct me (or point to a page where I can found this information) how can freeswitch be used as a sip client. I would like to test a server transporting SIP packets over TCP and was reported on this mailing list that it is possible to do so with FS. thanks, Tzury

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SIPGATE Problem
by NOx-WHV 18 Aug '09

18 Aug '09

Hi, i have a problem using my freeswitch with a sipgate account. The gateway entry is o.k.. Freeswitch try a call, sent an invite and sipgate answer with "403" (forbidden). I guess the string is incorrect. I modify the "effective_caller_id_number" to accountnumber(a)sipgate.de but the trace show the strting from: sip:accountnumber@sipgate.de@my-ip-address How can i modify the string to sip:accountnumber@sipgate.de without my ip-address. -- View this message in context: http://www.nabble.com/SIPGATE-Problem-tp25005636p25005636.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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"mute" channel programmatically with mod_event_socket
by Artem Shiyanov 18 Aug '09

18 Aug '09

Hello all! I'm trying to implement "mute" feature with mod_event_socket: I want programmatically mute/unmute some channel in a call.. And I don't see any other ways except to use conference room with special rule "mute". Anybody knows the better way? Thanks, Artem

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Setting WAV File as rinback
by Max Bridgewater 18 Aug '09

18 Aug '09

Hi, I'm trying to have an audio as ringback (WAV, 8khz, mono) when originating a call. Unfortunately, it doesn't seem to work; the RINGING teletone is being used instead of my audio. I have debug level enabled on Freeswitch but i don't find anything suspicious. is there anything specific i should be taking into consideration? Thanks, max.

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G729 transcoding workaround
by Justin Miller 18 Aug '09

18 Aug '09

To overcome the G729 transcoding issue with voicemail, I'm using an Audiocodes Mediant 1000 for transcoding. Our SIP trunk provider and all of our phones use G729 exclusively. When a call needs to go to voicemail, the call is bridged to the M1000, which transcodes to G711, and returns the call to Freeswitch on another port (5090). This seems to be working well. I'm now working on getting the IVR working so that I can start using the PBX functionality of Freeswitch for our office. When a call needs to hit the IVR, it's bridged to the M1000 the same as voicemail. I hear the custom menu I've created, and I can choose options successfully, except that the call is now operating in G711. When I bridge to a phone, I get the transcoding error. One option would be to transcode through the M1000 yet again, but this would take up 2 more DSP sessions in the M1000, and I would be running a total of 6 call legs in Freeswitch for this one call. Is there a way to end the transcoded call legs, and bridge to the phone from the original call leg? This would free up the M1000, and just seems like a better way to do things. Thank you Justin

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Eavesdrop getting killed after being answered
by Lars Zeb 18 Aug '09

18 Aug '09

I used to be able to dial 88+extension to eavesdrop, but now it is killed right after the call is answered by the extension. Can anyone tell me what I have done wrong? I am running version 14534 on Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux. http://pastebin.freeswitch.org/10025 Thanks Lars

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JAVA ESL
by Jean-Marc Hyppolite 17 Aug '09

17 Aug '09

Hello, I built libesl and JAVA mod. (make and make javamod). But when I try to run a JAVA script with the following code ESLconnection connection = new ESLconnection("127.0.0.1", "9000", ""); ESLevent events = connection.getInfo(); System.out.println(events.toString()); I get the following error message: Exception in thread "main" java.lang.UnsatisfiedLinkError: /usr/lib/libesl.so: /usr/lib/libesl.so: undefined symbol: __gxx_personality_v0 at java.lang.ClassLoader$NativeLibrary.load(Native Method) at java.lang.ClassLoader.loadLibrary0(ClassLoader.java:1778) at java.lang.ClassLoader.loadLibrary(ClassLoader.java:1703) at java.lang.Runtime.loadLibrary0(Runtime.java:823) at java.lang.System.loadLibrary(System.java:1030) at ivr.IVRServer.<clinit>(IVRServer.java:18) Any help would be appreciated. Thanks. __________________________________________________________________ Ask a question on any topic and get answers from real people. Go to Yahoo! Answers and share what you know at http://ca.answers.yahoo.com

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Cluecon 2009
by David Knell 17 Aug '09

17 Aug '09

Just a quick note to say thanks to Cluecon's organisers for putting together such a useful, informative and packed three days. I've come away with a head full of ideas, a bunch of new contacts and a collection of things to do; I'd thoroughly recommend that anyone interested in IP telephony blocks out the first week of August 2010, right now..! Cheers -- Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave(a)3c.co.uk W: http://www.3c.co.uk

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ext sip & rtp IP trouble
by kokoska rokoska 17 Aug '09

17 Aug '09

Hi all, I have just upgarded my (a little bit old :-) FreeSWITCH to current trunk and now i have troubles with IP addresses in SIP & SDP, while before upgrade everything work fine... My FreeSWITCH is behind NAT and all relevant sip profiles have set: <param name="ext-rtp-ip" value="1.2.3.4"/> <param name="ext-sip-ip" value="1.2.3.4"/> But when I send INVITE thru the profile to the gateway associated with it i see local IP in "Via" header and in SDP "m" line too. BTW: The "Contact" header is populated with "external" IP (1.2.3.4) Could you, please, point me how could I modify IP in Via and in SDP? Thank you. Best regards, kokoska.rokoska

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