- FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH how to configure for Multi-Tenant?
by Thomas Lee 04 Apr '13

04 Apr '13

FreeSWITCH how to configure for Multi-Tenant? Thanks Regards, Thomas Lee -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-how-to-configure-f… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Is disable-rtp-auto-adjust broken?
by Josh Nankin 04 Apr '13

04 Apr '13

I keep seeing lines like the following in my logs: 2013-04-03 19:14:09.252741 [INFO] switch_rtp.c:3684 Auto Changing port from xx.xx.xx.xx:25682 to xx.xx.xx.xx:25622 I have tried using the disable-rtp-auto-adjust option in my sip profile, as well as specifying it in my dialplan as well using the respective variable. Nothing seems to stop this. I'm trying to disable auto changing ports because when receiving a fax Freeswitch claims to be transmitting a CED message in the logs, but this message is not showing up in packet capture flows when viewing calls in Wireshark. My thoughts are currently that this is because freeswitch is not sending packets to the port requested by my SIP provider, and instead sending it to some unexpected port.

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Can be modified switch_m_sdp?
by Adrian Fuentes 04 Apr '13

04 Apr '13

Hi all Is it possible to replace the sdp c = IN IP4 172.25.1.9 through 172.25.1.9 c = IN IP4? variable_switch_m_sdp: [v=0 o=- 13649160524104272291 13649160524104272291 IN IP4 10.5.86.99 s=it-ssw c=IN IP4 172.25.1.9 t=0 0 m=audio 6150 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=silenceSupp:off - - - - ] Regards Adrian Fuentes

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Outbound blocking code
by Bala Murugan Mahendran 03 Apr '13

03 Apr '13

Looking to block few countries (or) need to allow few countries to make outbound call, is there a wiki examples for this? Thanks, Bala

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RFC4904 Trunkgroup Identity
by Kevin Kennedy 03 Apr '13

03 Apr '13

I am new to freeswitch as far as testing it, but have been on the user-list for a long time. I have searched through my archive of emails as well as searched on google for any answer that I can find on this. I am looking for the configuration options for RFC4904, trunk-group identity. This is where you can set Freeswitch up to send unscreened calls through your trunkgroup using the tgrp and trunk-context that is in the registration for every outbound call. Any help would be appreciated.

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Come join the conference call!
by Michael Collins 03 Apr '13

03 Apr '13

We have several items to discuss, so please join us! http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_03 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Getting FreeTDM originate status
by Ashish gautam 03 Apr '13

03 Apr '13

Hi, I have connected a PRI line to FreeSWITCH working with mod_freetdm enabled. Calls are going properly. Now, what I want is to get the originate status of each call i.e. whether the call was connected, failed, user busy or whatever the status is. How can I get this status from FreeSWTICH or FreeTDM? I am generating outgoing calls through event socket which receives originate request from a Perl script running on the same server. Please help and throw some light on this. -- Regards, Ashish

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choppy audio
by Jimmy Chang 03 Apr '13

03 Apr '13

Hi, I have two softphone clients commnunicating by G729. The audio quality of direct call between them is perfect. But when talking via FS, I got choppy audio in agent side(client side remains perfect). I had set bypass_media_after_bridge=true in dialplan, not work, and the RTP traffic is still through FS. Any advice? Thanks. Jimmy

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Can we add parameters to Request URI in SIP INVITE Message? FreeSWITCH how to get those parameters?
by Thomas Lee 03 Apr '13

03 Apr '13

Can we add parameters to Request URI in SIP INVITE Message? FreeSWITCH how to get those parameters? Thanks Regards, Thomas Lee -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-we-add-parameters-to-Requ… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Dynamic codec switching
by Daniel Ivanov 03 Apr '13

03 Apr '13

I remember fairly well that there is dynamic codec negotiation in fs from a big thread here on the list. My question is can i use this to overcome transcoding difficulties for ringback. I have g729 licences on the box, but prefer amr for mobile connections. Can i send early media in g729 and then switch to amr for actual call?

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