April 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Re: [Freeswitch-users] Recording incoming calls
by Clive Lansink 02 Apr '13

02 Apr '13

Thanks Michael. However I have to admit I'm a little out of my depth here. I know what works but I don't completely know the detail of why. Can you give me a syntax of what you think I should try? Thank you. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: Michael Collins <msc(a)freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Recording incoming calls Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: Mon, 1 Apr 2013 12:57:02 -0700 Clive, I think the execute_on_answer is the way to go. However I believe you need to set that on the b-leg, so either include it as part of the bridge data or use the export app instead of set. Let us know how it goes. -MC On Sun, Mar 31, 2013 at 2:06 PM, Clive Lansink <clive(a)lansink.co.nz> wrote: > Hello everyone > > I want to be able to record incoming calls. I had this all working using > record_session, until someone told me that when they phone in they no > longer hear any ringing before the phone is answered. Now I'm trying to set > things up so record_session works but without interfeering with the ringing > sound heard by the incoming caller. > > Incoming calls are received in the public context and then transferred to > extension 2000 in the default xml context. Then we have a local extension > in the default.xml dial plan that bridges to the incoming call group. If > the call is not answered within a certain timeframe, it transfers to > extension 1999 for voice mail - ie we don't have voice mail on each > extension. The definition of this extension is as follows: > <extension name="group_dial_IncomingCall"> > <condition field="destination_number" expression="^2000$"> > <action application="set" data="hangup_after_bridge=true"/> > <action application="set" data="continue_on_fail=true"/> > <action application="hash" > data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> > <action application="bridge" > data="{leg_timeout=18,ignore_early_media=true}${group_call(IncomingCall@ > ${domain_name})}"/> > <action application="transfer" data="1999 xml default"/> > </condition> > </extension> > > My first attempt to add record_session was as follows: > <action application="set" data="RECORD_TITLE=Recording > ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> > <action application="set" > data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> > <action application="set" data="RECORD_STEREO=true"/> > <action application="set" data="recording_follow_transfer=true"/> > <action application="record_session" > data="p:/media/phonecalls/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > This works except that the incoming caller hears no ringing sound while > our phones are ringing. In the log I see something like "Pre-Answer > sofia/external/021663999(a)27.111.14.67!" That is only there when the > record_session stuff is included so I presume the problem is that > record_session is somehow engaging with the call and the incoming caller > gets what amounts to an answered call, even though we haven't actually > answered it. > > Then I added > <action application="set" data="media_bug_answer_req=true"/> > but although the recording now starts at the time we answer the call, the > pre-answer thing is stil there and the incoming caller hears no ringing > sound. > > So then instead of executing record_session then and there as an > application, I tried > <action application="set" data="execute_on_answer record_session > p:/media/phonecalls/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > I thought this might execute the record_session application when we answer > the call but it hasn't worked. > > Does anyone have any suggestions? > > > Clive Lansink > Email: Clive(a)Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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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Weirdness
by Blake Priddy 02 Apr '13

02 Apr '13

Why would everyone of my extensions be able to call a certain extension number such as (3003) but when a particular user calls that number FS tells that user (3006) that 3003 is busy? 2013-04-02 08:18:56.104087 [DEBUG] sofia.c:5603 Channel sofia/internal/ sip:3003@10.25.190.8:1024 entering state [calling][0] 2013-04-02 08:18:56.254094 [DEBUG] sofia.c:5603 Channel sofia/internal/ sip:3003@10.25.190.8:1024 entering state [terminated][486] 2013-04-02 08:18:56.254094 [DEBUG] switch_channel.c:2994 (sofia/internal/ sip:3003@10.25.190.8:1024) Callstate Change RINGING -> HANGUP 2013-04-02 08:18:56.254094 [NOTICE] sofia.c:6387 Hangup sofia/internal/ sip:3003@10.25.190.8:1024 [CS_CONSUME_MEDIA] [USER_BUSY] 2013-04-02 08:18:56.254094 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:3003@10.25.190.8:1024) Running State Change CS_HANGUP 2013-04-02 08:18:56.254094 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/sip:3003@10.25.190.8:1024 [KILL] 2013-04-02 08:18:56.254094 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:3003@10.25.190.8:1024 [BREAK] 2013-04-02 08:18:56.254094 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:3003@10.25.190.8:1024) State HANGUP 2013-04-02 08:18:56.254094 [DEBUG] mod_sofia.c:503 Channel sofia/internal/ sip:3003@10.25.190.8:1024 hanging up, cause: USER_BUSY 2013-04-02 08:18:56.274104 [DEBUG] switch_ivr_originate.c:3531 Originate Resulted in Error Cause: 17 [USER_BUSY] 2013-04-02 08:18:56.274104 [NOTICE] switch_ivr_originate.c:2608 Cannot create outgoing channel of type [user] cause: [USER_BUSY] 2013-04-02 08:18:56.274104 [DEBUG] switch_ivr_originate.c:3531 Originate Resulted in Error Cause: 17 [USER_BUSY] 2013-04-02 08:18:56.274104 [INFO] mod_dptools.c:3052 Originate Failed. Cause: USER_BUSY EXECUTE sofia/internal/3006(a)pbx.bryantschools.org answer() -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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Yum RPM Repo Metadat Corrupt
by Scott Ocken 02 Apr '13

02 Apr '13

It looks like the Yum RPM Repo for Free Switch is corrupt. I am getting: http://files.freeswitch.org/yum/6/x86_64/repodata/filelists.xml.gz: [Errno -1] Metadata file does not match checksum Trying other mirror. http://files.freeswitch.org/yum/6/x86_64/repodata/primary.xml.gz: [Errno 14] Downloaded more than max size for http://files.freeswitch.org/yum/6/x86_64/repodata/primary.xml.gz: 101360 > 91756 Trying other mirror. Error: failure: repodata/primary.xml.gz from freeswitch: [Errno 256] No more mirrors to try. Thanks Scott

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