April 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

DID Provider
by Saeed Ahmed 28 Apr '09

28 Apr '09

Hi All, Could you please refer me to a best DID provider which also works perfectly with FreeSWITCH. I would need 50 channels per DID number. Thanks.

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FreeSWITCH In Blogosphere
by Michael Collins 28 Apr '09

28 Apr '09

I just wanted to let everyone know that there are a few blog posts out there that you FreeSWITCHers might find interesting. Please check out these two stories on the main FreeSWITCH page: Another Interesting Use For FreeSWITCH: OpenSim<http://www.freeswitch.org/node/175> DIDX Interview With Anthony Minessale and Brian West<http://DIDX%20Interview%20With%20Anthony%20Minessale%20and%20Brian%20West> Be sure to leave comments where appropriate. I will followup with digg links shortly. -MC

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Low rtp-timeout-sec hangs up call in ringing state - expected behaviour?
by Mikael Aleksander Bjerkeland 28 Apr '09

28 Apr '09

Hi, I have been testing inbound calls to a Nokia phone with handover to a cellphone number if I get MEDIA_TIMEOUT on the B leg of the call, and had to set rtp-timeout to a very low 6 seconds in order to get "fast" handover. This introduces an interesting side-effect that hangs up calls even in the ringing state after 6 seconds. Is this the desired behaviour of rtp-timeout-sec? My initial guess was that rtp-timeout-sec should only be valid for established calls where the two endpoints have exchanged rtp at some point but have stopped exchanging media. As far as I know a phone call in ringing state has not shared any RTP with the other endpoint until it gets early media or is answered. Should rtp-timeout-sec really be valid even when ringing? It seems to me that setting rtp-timeout-sec to 60 seconds would add an absolute time limit on ringing phone calls to 60 seconds, which I believe is not the actual purpose of this limit. Could anyone please share their thoughts on this matter? Thanks, Mikael

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Anybody tried Nokia E71 (symbian S60 3rd) with Pjsip and TLS/SRTP?
by Peter P GMX 28 Apr '09

28 Apr '09

Hallo, has anybody tried successfully to setup a Nokia E71 (or similar symbian S60 3rd phone) with Pjsip and TLS/SRTP? TLS seems to work but what about the SRTP part? Best regards Peter

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Sofia doubts
by Arsen Chaloyan 28 Apr '09

28 Apr '09

What I want to discuss concerns mostly Sofia-SIP, but not primary FreeSWITCH. Surely it would be better to use Sofia's list instead and I already did it, but have got no response so far. Overall Sofia-SIP is a good library, which works reliable enough on Linux, but Windows port causes me some doubts. Does anybody use FreeSWITCH/Sofia-SIP in production ready environment on Windows? I know it's working, but how far it has been tested. Just to be clear, the issues I've observed concerns UniMRCP/Sofia-SIP 1.12.10 and I just wanted to know how reliable it works on your end. http://sourceforge.net/mailarchive/forum.php?thread_name=708583.92229.qm%40… http://sourceforge.net/mailarchive/forum.php?thread_name=1ae1714d0903290440… Thanks, Arsen. www.unimrcp.org

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Re: [Freeswitch-users] [Freeswitch-dev] Compiling the FreeSwitch on Windows XP
by David A. Horner 28 Apr '09

28 Apr '09

Start with installing the windows sdk..... error C2143: syntax error : missing ']' before 'constant' C:\Program Files\Microsoft SDKs\Windows\v6.0A\include\ras.h http://msdn.microsoft.com/en-us/windows/bb980924.aspx --dave http://dave.thehorners.com/tech-talk/programming On Tue, Apr 28, 2009 at 4:15 AM, Santosh <santosh_tripathi(a)datamatics.com> wrote: > Hi, > I am compiling the FreeSwitch 1.0.3 on Windows machine and getting hundreds > of error.I guess this has something to do with Visual studio Settings.Can > you please help me with this.I am attaching the Error file also(in which > there are some missing files too).Can you send me a working FreeSwitch code > for windows? > > Regards, > Santosh > > > > > Disclaimer: The information contained in this e-mail and attachments if any > are privileged and confidential and are intended for the individual(s) or > entity(ies) named in this e-mail. If the reader or recipient is not the > intended recipient, or employee or agent responsible for delivering to the > intended recipient, you are hereby notified that dissemination, distribution > or copying of this communication or attachments thereof is strictly > prohibited. IF YOU RECEIVE this communication in error, please immediately > notify the sender and return the original message. > > > > _______________________________________________ > Freeswitch-dev mailing list > Freeswitch-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > >

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Internal.xml using Public context/dialplan?
by Fred-145 28 Apr '09

28 Apr '09

Hello I'm going through the various configuration files to figure out how they work together, and noticed the following references: sofia.conf.xml -> sip_profiles/internal.xml -> "public" context. 1. Am I right in understanding that a context is a file in the dialplan/ subdirectory? 2. Shouldn't the internal.xml SIP profile refer to the "default" (internal) context instead of "public" (external)? Thank you. -- View this message in context: http://www.nabble.com/Internal.xml-using-Public-context-dialplan--tp2317544… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Adding Spanish support to say
by Kristian Kielhofner 28 Apr '09

28 Apr '09

Hello everyone, I'm trying to add Spanish support to say. I'm using something like: <include> <language name="es" sound-path="$${base_dir}/sounds/es/mx/asterisk" tts-engine="cepstral" tts-voice="callie"> <X-PRE-PROCESS cmd="include" data="demo/*.xml"/>   <X-PRE-PROCESS cmd="include" data="vm/sounds.xml"/>  </language> </include> in conf/lang/es which is included by freeswitch.conf: <X-PRE-PROCESS cmd="include" data="lang/es/*.xml"/> ..right after English. Yet I continue to get [ERR] switch_ivr.c:2014 switch_ivr_say() Invalid SAY Interface [es]! Whenever trying to use say: <action application="say" data="es name_spelled pronounced ${caller_id_number}"/> What am I missing? Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

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Problem with a hang up event on transfer.
by mszlazak＠aol.com 27 Apr '09

27 Apr '09

Using 13157 and some prior builds, I'm having a problem with the hang up hook activating on a transfer at the end of the script. The hook is suppose to fire with hang ups except when the transfer occurs and I thought setAutoHangup(false) would prevent this but it doesn't. Here is how things are set up. session.setHangupHook(on_hangup); .... .... session.setAutoHangup(false); session.execute("transfer", "NOTIFY_ME XML default"); session = undefined; If setAutoHangup(false) is working correctly then is there a method to "unset" the hang up hook before the transfer? Mark.

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FS 13157 fsctl shutdown restart sequence fails ...
by mszlazak＠aol.com 27 Apr '09

27 Apr '09

In FS trunk 13157, fsctl shutdown restart fails to work as in the past. I get the complete shutdown without the restart. Mark.

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