May 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

uuid_transfer gets break
by Gopalakrishnan A.N 29 May '09

29 May '09

Hi, I had some discussion with the IRC regarding the uuid_transfer gets hang-up where the call is originated via javascript thru event socket. I was suggested to install latest SVN trunk. I did that again the same issue, the log is attached with here http://pastebin.freeswitch.org/9103 My call flow like this, 1. api jsrun fils.js <arg1> <arg2> <arg3> 2. capture the uuid 3. api uuid_transfer <uuid> -both <ext no / conference no> Both the leg gets hangedup. Someone can assist me where I am wrong. -- Thank you with regards, Gopal,

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Freeswitch with APR
by Gopalakrishnan A.N 29 May '09

29 May '09

Hi, I saw the apache portable runtime is included in freeswitch. So far I understand that using APR will give good performance. Am I correct? or it has been used for some other scenarios like even socket or dialplan? -- Thank you with regards, Gopal,

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FW: Something broken in RTP in latest trunk (r13502)?
by Peter Olsson 29 May '09

29 May '09

Sorry for missing this in my last post, but I'm using sofia for all calls. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Peter Olsson Skickat: den 29 maj 2009 12:31 Till: 'freeswitch-users(a)lists.freeswitch.org' Ämne: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? After using the latest trunk revisions I get no audio anymore. The last working build I have is about 5 days ago. I havn't upgraded until today, so I don't know exactly when this happened. I've noticed quite a few changes on the RTP stack, beacuse of the implementation om ZRTP, and I guess it's somewhere around this time when it happened. How to continue debugging on this issue? I have both a working version of FS (compiled 5 days ago), and a broken one (compiled today), so I can test this very easily, and everything is on a non live server. The conf-dir is the same between the revisions. The calls I'm trying to do is both directly to FS (voicemail or similar applications), and aslo calls to another SIP-trunk, to PSTN (media stream is sent through FS). Regards, Peter !DSPAM:4a1fbb5432933803719169!

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gtalk text chat
by Mark Campbell-Smith 29 May '09

29 May '09

Hi! Is it possible to configure freeswitch and mod_dingaling so that it sends a text chat to a gtalk client? I would plan to use this for certain debugging purposes. Thanks! /Mark

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Default IVR action
by Nandy Dagondon 29 May '09

29 May '09

hi to all, i'm looking for a default action in the IVR in case the caller doesn't press any key. is this option available? with this option, we can add this prompt "... please stay on the line to be connected.". i know this can be done using scripts but it's better to have this feature on the app itself. rgds, -nandy =============================== LanVox Systems Lapulapu City, Philippines 6015 Mobile: +63-920-6373450 Phone: +63-32-3401807 USA: +1-360-8122281 http://sites.google.com/site/lanvoxphils

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Re: [Freeswitch-users] The calls are dropped during register
by Diego Toro 29 May '09

29 May '09

thank you Brian, my notes on http://jira.freeswitch.org/browse/SFSIP-143, I have hardware avaible for testing Diego --- On Thu, 5/28/09, Brian West <brian(a)freeswitch.org> wrote: From: Brian West <brian(a)freeswitch.org> Subject: Re: [Freeswitch-users] The calls are dropped during register To: freeswitch-users(a)lists.freeswitch.org Date: Thursday, May 28, 2009, 8:04 PM Please add all these notes to the jira. I don't mind helping but its not on the top priority list right now as its only effecting two people with odd hardware. I need to know if others are seeing this behavior. /b On May 28, 2009, at 8:02 PM, Diego Toro wrote: Setting expire-seconds greater than or equal to 120 the register is lost. thank you Diego Brian West brian(a)freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -----Inline Attachment Follows----- _______________________________________________ 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

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Re: [Freeswitch-users] The calls are dropped during register
by Diego Toro 29 May '09

29 May '09

Setting expire-seconds greater than or equal to 120 the register is lost. thank you Diego --- On Thu, 5/28/09, Brian West <brian(a)freeswitch.org> wrote: From: Brian West <brian(a)freeswitch.org> Subject: Re: [Freeswitch-users] The calls are dropped during register To: freeswitch-users(a)lists.freeswitch.org Date: Thursday, May 28, 2009, 7:54 PM Don't set your interval lower than 120 like you said.. if it hurts don't pick at it. /b On May 28, 2009, at 7:49 PM, Diego Toro wrote: Greetings The calls are dropped when FS try to register whith a pbx, I see mutiples requests of register from FS to pbx, although the register isn't lost, the calls are dropped. I have console trace: http://pastebin.freeswitch.org/9162 I appreciate any help Diego Brian West brian(a)freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -----Inline Attachment Follows----- _______________________________________________ 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

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The calls are dropped during register
by Diego Toro 29 May '09

29 May '09

Greetings The calls are dropped when FS try to register whith a pbx, I see mutiples requests of register from FS to pbx, although the register isn't lost, the calls are dropped. I have console trace: http://pastebin.freeswitch.org/9162 I appreciate any help Diego

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FS in Amazon EC2 for production?
by Ing. Edwin Villarreal 28 May '09

28 May '09

Hello my friends. Has anyone used the EC2 for production? Tests? I'm wondering if it would be "better" to have a FS system in the cloud for carrier-to-carrier connections. Any ideas will be appreciated Thanks 2 u all Edwin Villarreal World Net Commerce SA CV WNC Telecom

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fax@pstn with freeswitch
by Uwe Kastens 28 May '09

28 May '09

Hello list, Since I tested the last year lot of T38 implementation with cpe, asterisk etc.pp. with lots of problems, I would like to have a solution to handle the fax issue with pstn. How is the mod_fax integrated with freeswitch? Is there a kind of "softmodem" needed, or could fax directly handled on a circuit of a sangoma E1 card? I would like to build a kind of routing device which could handle DDI correctly and is able to hold calls on TDM for fax. Example: I have a kind of numbers like 12345-[0-6] The user has a fax number on 12345-523 and 12345-455. Then I would like to handle 12345523 => fax-mod@tdm 12345455 => fax-mod@tdm 12345[0-6] => SIP BR Uwe -- kiste lat: 54.322684, lon: 10.13586

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