- FreeSWITCH-users - FreeSWITCH Mailing Lists

Per User Music on Hold
by Matt Putnam 25 Mar '10

25 Mar '10

Is there a way to change music on hold per user instead of per channel? I have one user that I would like to have different hold music than the default. I can get it to work outbound by setting it on the channel but inbound I cant seem to find a way to change its always the default specified in vars.xml.

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Media traffic to bypass FreeSWITCH and to be exchanged by peers only
by Vitalii Colosov 25 Mar '10

25 Mar '10

Hi, Very newbie question: Currently both SIP and RTP traffic passes thru the FreeSWITCH server. Client A -->SIP and RTP -->FS server-->Client B Is it possible that only SIP traffic will pass thru FS, and media will be exchanged between Client A and Client B directly? Client A -->SIP -->FS server-->Client B Client A -->RTP -->Client B How can I configure this? Sure both client A and B will use same codec, otherwise FS Server is a must for transcoding, right? Or is a must for anything else... Thank you, vIT

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international prefix
by Madovsky 25 Mar '10

25 Mar '10

I try to manage trunks that allow different international prefix. for example, trunk A allows "011" and trunk B "00", and for domestic calls trunk B allows no prefix. it's a problem when I use mod distributor since I can't gues which trunk will be used for the bridge. I'm not sure that lcr is a good solution for this problem as it makes more DB overload. Any idea ? Thanks

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send fake UDP packets
by Madovsky 25 Mar '10

25 Mar '10

I use an audio stream relayed by a client that has no audio DSP card. so when I stop to talk, no UDP audio packet is sent to FS. Do you know a trick to send fake UDP packet to keep the call alive ? (I need to keep rtp-timeout short to hang up leg B when leg A hangs up) Thanks F

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Re: [Freeswitch-users] Guidance on debugging audio glitches
by Chris Fowler 25 Mar '10

25 Mar '10

Forgot to mention : Running trunk - build 17104 From: Chris Fowler Sent: Thursday, March 25, 2010 12:48 PM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: Guidance on debugging audio glitches Hi, Here's the problem I'm "hearing": Test Case #1 x1001 calls x1002 - Audio is rock solid Test Case #2 x1001 calls x3001 (Conference Bridge) x1002 calls x3001 (Conference Bridge) - Audio is mostly perfect. However, occasional "glitches" in the audio can be heard Notes: - Bypass media is switched off. FreeSWITCH routes audio for both the bridged, and the conferenced call - rtp-autoflush-during-bridge is "true" on Sofia profile internal.xml - I enabled the debug trace for auto-flush in switch_rtp.c (line 1895) and see occasional auto flush of rtp packets for *both* test cases. The glitches don't coincide with the flushing. In fact with flushing you can't hear when data gets flushed (2x20ms packets). Environment - CentOS 5.4 - Tested on AWS EC2 Small, Medium, Large instances. On Medium and Large CPU affinity for FreeSWITCH set to the 2nd core. Questions: - What's different between bridging a call and a conference call run through mod_conference in terms of audio processing? - Thoughts on how to debug this? Thanks, Chris.

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variables scope
by Madovsky 25 Mar '10

25 Mar '10

Hello, what is the scope of variables set in the gateways config ? also is there any wiki that explains the scope of different vars ? Thanks

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How to force a Extension use g729 only
by Aloysius Lloyd 25 Mar '10

25 Mar '10

Hi All, I have the following setup [IP Phones] ----[Router] --------------[FreeSWITCH] ------ [ITSP] How to force a extension use g729 only. I could not find any parameter to set for each extension. In Asterisk world I can do disallow=all allow=g729 Thanks, Lloyd

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variables in gateway profile
by Madovsky 25 Mar '10

25 Mar '10

Is it possible to create <variable> in gateway profiles like users profiles ? Thanks Franck

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IRC Day!
by Michael Collins 25 Mar '10

25 Mar '10

Hello all FreeSWITCHers! We invite everyone around the world to join us in #freeswitch on irc.freenode.net. We want everyone to be online today, even if you just leave your IRC client running in the background. Also, if you don't have an IRC client handy then go to http://www.freeswitch.org/ and use the web login located on the right-hand side bar (below the features list). You'll need a Java-enable browser. Everyone have a great day! -Michael

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Confusion - Please Help
by Ken Fulmer 25 Mar '10

25 Mar '10

We need the internal sipX box to register phones and send / receive PSTN calls through the FS SBC. I don't understand how we need to setup SIP Profiles in FS. We basically need one gateway to an ITSP and one gateway to our sipX PBX. Is this done in different places? Thanks, Ken Fulmer

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