March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SILK Codec
by Brian West 31 Mar '10

31 Mar '10

We now have mod_silk in tree it has only been tested on Linux so far. The silk library itself was released in the IETF draft only an hour and twenty minutes ago. I wrote a perl script to extract the source from the draft. Checked in the lib and the codec module which I had written to the binary lib I had a few months ago. It has more work to do... but its there if anyone wants to chip in and libtoolize the library, revamp the build system and assist in testing the codec on multiple platforms. Thanks, Brian

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SIP Call Trace
by Ken Fulmer 31 Mar '10

31 Mar '10

Perhaps I've missed it, but can someone explain how to capture SIP and SDP messages for troubleshooting? Thanks, Ken Fulmer

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Adding Record-Route Header
by Suneel Papineni 31 Mar '10

31 Mar '10

Hi, How can I add Record-Route (pointing to another FS acting as Proxy server) for the calls handled by to FSComm client. Also can I set this while making a call or accepting a call using event socket commands (pa answer / pa call xxxx). Thanks & Regards Suneel ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN *************************************************************************

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bridge from-user
by Justin B Newman 31 Mar '10

31 Mar '10

I have tried a number of variations on: <action application="bridge" data="[from-user=abc,sip_auth_username=abc,sip_auth_password=123]sofia/external/$1(a)voip.mydomain.com"/> So far, to no avail. In all cases, the From does not contain "abc", but rather, the Caller-ID data. FreeSWITCH Version 1.0.5-20100329-0400 (17128M) Is it possible to modify From when placing an outbound call in absence of a gateway? -jbn

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debugging no audio I/O
by Neil Patel 31 Mar '10

31 Mar '10

Hi all, I'm stuck trying to figure out why my lua app does not output any audio or detect any DTMF input. This app has been installed and running successfully on another server. The lua code in question is: read(absolute_path_to_wav, 0); Logs show: 2010-03-29 20:12:15.596856 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16@8000hz 1 channels 20ms 2010-03-29 20:12:15.596856 [DEBUG] switch_ivr_play_say.c:1444 done playing file DTMF input is not being recognized, nothing is showing up in the logs. App is being accessed with SIP from a remote client. I'm not really sure how to debug this, so any pointers are appreciated. This is FS from trunk. Thanks, Neil

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Calling Phone Behind Netgear Router (with NAT)
by Jerry Richards 31 Mar '10

31 Mar '10

We are trying call a phone that resides behind a Netgear router (with NAT). The Bria softphone registers and we can make calls from the softphone, but we cannot make calls to the softphone. WAN ------------ public IP -------- 10.10.10.xx ---------------- |Freeswitch|<---------->|Router|<----------->|Bria softphone| ------------ -------- ---------------- Does anyone know of good documentation/example showing this configuration? Best Regards, Jerry

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Re: [Freeswitch-users] Some question about mod_fifo ??
by Nguyễn Mạnh Hùng 31 Mar '10

31 Mar '10

Hi Mike and Seven Du. Thanks to yours help. I known the mechanism of mod_fifo. >> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175… . What a pity, It can't solve this problem. I can't use freeswitch for my call center. Hope new version can solve this !!!

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I need some comment for Priority Call
by Dome Charoenyost 31 Mar '10

31 Mar '10

Dear All, I'm planing to make priority call in dial plan. it's mean if we have maximum 30 concurrent port for dial out. and use all port. when some extention make call (VIP call) dial plan hangup some longest call and give port for VIP. any idea ? Dome C.

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Regular Expression for match ip address
by Dome Charoenyost 31 Mar '10

31 Mar '10

Dear All, I want to autorize incoming call to public context by IP by <condition field="network_addr" expression="^192\.168\.1\.1$"/> but some time i want to allow 1 Class C or some range of IP. How can do that ? BG Dome C.

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Variable "default_language" has no effect on javascript
by eliimap 31 Mar '10

31 Mar '10

Hi I set the variable "default_language" to be "ru" and then I run a js to play a file .... session.streamFile("conference/ronf-kicked.wav", ""); .... I got an error message 2010-03-25 10:23:46.430391 [ERR] mod_sndfile.c:194 Error Opening File [/freeswitch-1.0.5-20100314-0400/sounds/en/us/callie/conference/ronf-kicked.wav] [System error : No such file or directory.] that indicates that mod_sndfile is not checking this value but always using en Please check Eli

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