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Call Transfer Question
by DJB 07 Sep '09

07 Sep '09

I am not sure if my question did not get post correctly earlier. I wonder whether anyone can give me any recommendations. Here is the call flow: I call from the PSTN (A party) into my Polycom phone (B-party) which is registered to FreeSwtich. The Freeswtich is setup not to route media as I have an SBC acting as a mirror proxy that will do all the NAT and media routing. The inbound call is setup fine and there is two way voice. I then blind transfer from the Polycom to my Cell phone. I see the polycom send a SIP refer to Freeswitch and it sends a 202 accepted fine and that leg between the Polycom (B party) and the A party is torn down fine like its supposed to be. The Freeswitch places the outbound call (the number the call is transferring to C-party) and that call completes. However now there is one way audio between the A party and C party . I see RTP streaming back from the egress carrier where the call was transfered to so the A party can hear the C party but the C party cannot hear the A party . When I look at the SIP traces of the original inbound call from the A-party I see a SIP re-invite from free switch to place the call on hold (contains Freeswitch RTP address to I can hear hold music) while it is transferring the call and the A-party does hear on hold music from Freeswitch while the call is being transferred. However I do not see a second re-invite from freeswitch to pass the media IP it got from the egress leg back to the original inbound leg. If my inbound gateway does not get a re-invite from Freeswitch to redirect its media to the new RTP address of of the egress carrier it will not do so hence the one way voice. How do I get the Freeswitch to re-invite the original ingress leg once it gets the SIP 183 from the egress with the new RTP info ? Free switch is sending the first SIP re-invite to my inbound gateway with new media IP (IP of itself) so the A-party can hear on hold music , but does not send a second re-invite to my inbound gateway after it receives the new RTP address from the egress carrier for the call that was transferred back out. Thank you.

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Call Forwarding Question
by Nikolai Geordzhev 06 Sep '09

06 Sep '09

Hi, I`m trying to implement Call Forwarding in my FS setup. I set a user variable managing the type of forwarding (busy,no answer,unconditional) and the destination the phone is forwarded to: <user id='102'> <variables> <variable name='fwd_type' value='unconditional'/> <variable name='fwd_unconditional_number' value='103"/> </variables> </user> Then in my dialplan I check if the called user has ${fwd_type} set and if so I set the destination_number to ${fwd_unconditional_number}. After this operation I bridge the call to the new destination_number. The issue in this scenario is that I get a cdr in which A (101) calls C(103) and I need to have two cdrs, one from A(101) to B(102) and one from B(102) to C(103). Does anyone know how can this be implemented. 10x, NG

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Skypiax working but laggy
by Gonzalo Servat 06 Sep '09

06 Sep '09

Hi All, I'm just testing out mod_skypiax (great work Giovanni & co!) and while it's working and all, I find that when I call in from a Skype contact, it's /very/ laggy. I would say something on the skype end and I would hear it on the FS end quite a bit later. Funny thing is the audio going the other way has much faster response. I'm running FreeSWITCH Version 1.0.trunk (14772) on Ubuntu 9.04 Jaunty. Do you need more info? Thanks, Gonzalo

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Re: [Freeswitch-users] how to call mod_commands from within a lua script
by Michael S Collins 06 Sep '09

06 Sep '09

On Sep 5, 2009, at 8:25 PM, Adam Wilt <wiltingtree(a)gmail.com> wrote: > Hi, the documentation says that mod_commands is available from > within mod_lua. But when I try to access it like this: > > session:execute("uuid_broadcast",session_id .. " " .. filename .. " > both") > > I get: Invalid Application uuid_broadcast > > or > > session:execute("bgapi","uuid_broadcast " .. session_id .. " " .. > filename .. " both") > > I get: Invalid Application bgapi > > or > > session:uuid_broadcast(session_id .. " " .. filename .. " both") > > I get: attempt to call field 'uuid_broadcast' (a nil value) > > > > Can somebody please explain how to do this? > > Thanks! > Absolutely. Using session:execute is only for dialplan applications. To do API commands, including all the stuff in mod_commands, you need to create an API object: api = freeswitch.API() Then send a command and the variable will receive the results: reply = api:executeString("version") Check out this page for more examples: http://wiki.freeswitch.org/wiki/Make_API_calls_directly_from_Lua_code -MC > _______________________________________________ > 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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Conferencing setup with FS
by Ujjval Karihaloo 06 Sep '09

06 Sep '09

Hi , I cannot seem to find a Document online for setting up conferencingon FreeSwitch. Can someone point me to one? Thx.

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how to call mod_commands from within a lua script
by Adam Wilt 06 Sep '09

06 Sep '09

Hi, the documentation says that mod_commands is available from within mod_lua. But when I try to access it like this: session:execute("uuid_broadcast",session_id .. " " .. filename .. " both") I get: Invalid Application uuid_broadcast or session:execute("bgapi","uuid_broadcast " .. session_id .. " " .. filename .. " both") I get: Invalid Application bgapi or session:uuid_broadcast(session_id .. " " .. filename .. " both") I get: attempt to call field 'uuid_broadcast' (a nil value) Can somebody please explain how to do this? Thanks!

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chat api problem
by Juan Backson 05 Sep '09

05 Sep '09

Hi, I am not able to get the chat api to work. Any idea why the following api command is not working? freeswitch(a)localhost.localdomain> chat sip|180001(a)192.168.1.102|testing API CALL [chat(sip|180001(a)192.168.1.102|testing)] output: Invalid freeswitch(a)localhost.localdomain> thx, jb

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restart when convenient
by Christian Löschenkohl 05 Sep '09

05 Sep '09

hello i'm looking for a possibility to restart freeswitch like it is possible with asterisk. for me i tried to created a script that looks for open channels and if no channel is open it restarts freeswitch with the init script (not the most efficient way). i think i would be great if we would have a buildin function for this, i think such command would help with maintenance and not only for me. br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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auto expiration
by Woody Dickson 05 Sep '09

05 Sep '09

Hi, I would like to set up freeswitch to automatically expire a user registration if either NOTIFY or REGISTER is not received within certain time frame. Does anyone know how to do that? Thanks, Woody

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Monitoring FreeSWITCH Events via Telnet socket
by Tina Martinez 05 Sep '09

05 Sep '09

Thank you for the guidance towards the ESL lib, but I'm encountering some problems. ----> make perlmod When I attempt to execute the make perlmod, I currently get the following error: /usr/bin/ld: cannot find -ldb collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 Prior to this, I encountered a similar error (cannot find -lgdbm) where I had to create a link to fix: libgdbm.so -> /usr/lib64/libgdbm.so.2.0.0 I'm suspecting that my installation of Perl is not looking in the lib64 directory as it should? Unfortunately my Perl skills are not sufficient to be hacking around with this stuff, anyone with thoughts or suggestions?

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