June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

unallocated number error after integrating FS with Kamailio
by Sherif Omran 05 Jun '12

05 Jun '12

Hello Guys, I have Kamailio + FS running on the same server and integrated together according to this tutorial as SBC + Media Services http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc Issues: 1- 2 Sip softphones are registered. I can see them online in kamailio. When I call FS Echo service, it works. However, when I call the other softphone number, it gives unallocated number. 2- If user is unreachable, I should get to his voice mail. But I get also unknown Invalid application voice mail. Any body figures this error? Thanks Sherif Omran

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Doubts with att_xfer
by Antonio Modesto 05 Jun '12

05 Jun '12

Hi, I am trying to enable att_xfer for calls that come from pstn, these call are being routed to an internal registered SIP Phone. In asterisk I had those options "Tt" in the Dial application to enable transfer to the caller party and to the calling party. The question is that for internal call I want both legs be able to transfer to whoever they want, but for external calls I want just the internal leg be able to transfer calls. How can I do that? Regards.

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Re: [Freeswitch-users] Music on hold while I add someone to the conference
by Chris B. Ware 05 Jun '12

05 Jun '12

Thank you Michael, I'll test it. In the meanwhile I've found this workaround: <extension name="Add new OB call to conference"> <condition field="destination_number" expression="^ADD_THIRD$"> <action application="set" data="res=${bgapi uuid_broadcast ${uuid_B} local_stream://moh}" />  <action application="play_and_get_digits" data="4 11 1 5000 # ivr/ivr-enter_destination_telephone_number.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+"/> <action application="set" data="res=${bgapi uuid_break ${uuid_B} }" />  <action application="execute_extension" data="ADD_CALL_TO_CONF_${target_num}"/> </condition> </extension> uuid_broadcast and uuid_break do the trick!

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internal/external dialing between different domains
by Miha 05 Jun '12

05 Jun '12

Hello, what is the best way to dial between domains (multi tentat) on same FS? Is better to route it internal or external (sbc send it back and use public dialplan to send it to right domain)? What would be the best way to route it internally, so that dialplan would not change if I add more domains? Users are now registered like this: if domain is enterprise.test.com, user will be number.enterprise. Thanks! BR; Miha

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Multi-tenant dilplans
by Miha 05 Jun '12

05 Jun '12

Hi, I set multi tenat for two domains which is forking well. Now I would like to seperate dialplans so that domain A will use default dialplan in directory dialplan/a and domain B will use default dialplan in directory dialplan/b. I have tried few things but every time fs use default dialplan. Thanks for help! Regards, Miha

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How do you split bridged calls ?
by Gilbert Amar 04 Jun '12

04 Jun '12

Hello I am trying to split bridged calls. Using mod_event_socket I try this api uuid_transfer c1061126-a429-11e1-9da3-85575e903927 -both playback::phrase:AC_0003: It does not work as FS seems to really need an extension. the other option I could try is : api uuid_transfer c1061126-a429-11e1-9da3-85575e903927 -both park and then for each leg: sendmsg call-command: execute execute-app-name: playback execute-app-arg: phrase:AC_0003: Is there a better way ? I am using FreeSWITCH Version 1.0.head (git-fdaa155 2012-02-10 13-17-54 +0000) Thank you Gilbert

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Setting custom SIP headers on transferring call leg.
by Venkatesh 04 Jun '12

04 Jun '12

Hi ! Apologies if this question has already been answered. I am developing a simple IVR application using JS. The application simply answers the call; does a bunch of stuff. One of the things it does is to transfer the call to an extension. I want to be able to set some custom SIP headers on the "outbound" call leg. I tried using session.setVariable("variable_sip_h_X-My-Header", "my_value"), but don't see the header in the outbound invite. I suspect that this is acting on the session object of the incoming call and not on the outbound. Wanted to know if freeswitch provides this capability and if yes, the semantics for the same when used via JS. Any help would be great. Regards, Venkatesh

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How to play ivr on background?
by Mohammad Emran 04 Jun '12

04 Jun '12

Hi I would like play minute announce ivr during the process of call. Cause it will reduce connect time on client side. Pls advise how to do that. Sent from my iPhone

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Music on hold while I add someone to the conference
by Chris B. Ware 04 Jun '12

04 Jun '12

Hi, I'm testing example reported here: http://wiki.freeswitch.org/wiki/Conference_Add_Call_Example I'd like to modify extension "Add new OB call to conference" in order to play music on hold to the other conference members , while moderator digit a new member number to add. In my case we have a conference with A (moderator) and B (first member) and A try to add C. That's my conf: <extension name="Add new OB call to conference"> <condition field="destination_number" expression="^ADD_THIRD$"> <action application="set" data="res=${bgapi uuid_hold ${uuid_A} }" />  <action application="play_and_get_digits" data="4 11 1 5000 # ivr/ivr-enter_destination_telephone_number.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+"/> <action application="set" data="res=${bgapi uuid_hold off ${uuid_A} }" />  <action application="execute_extension" data="ADD_CALL_TO_CONF_${target_num}"/> </condition> </extension> Everything works except B can't hear any music. Where's the error? Thank you

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FreeSWITCH Conference Call June 6
by Michael Collins 04 Jun '12

04 Jun '12

Hello all! Just an update: We've rescheduled Darren's SIP 101 discussion for June 20th. Also, this Wednesday, June 6th yours truly will be doing an introduction to mod_httapi! We will be keeping it simple this week: just getting your server up and running, trying out the sample code, and maybe we'll do some what-if's from the audience. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_06 Keep in mind that mod_httapi allows for *your* FreeSWITCH server to call *my * web server to get dialplan instructions, so be sure to compile and enable mod_httapi <http://wiki.freeswitch.org/wiki/Mod_httapi> on your systems. Talk to you Wednesday! -Michael

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