January 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

api_hangup_hook not actually executing the command
by Scott Ellis 24 Jan '09

24 Jan '09

The on answer is fine, the api_hangup_hook gets to the point where it wants to execute, but then nothing happens. Any thoughts? 2009-01-21 03:08:00 [DEBUG] switch_channel.c:1773 switch_channel_perform_mark_answered() sofia/internal/Line_9(a)10.0.0.9 execute on answer: set_global(10.0.0.19_INCALL=true)2009-01-21 03:08:00 [DEBUG] mod_dptools.c:726 set_global_function() SET GLOBAL [10.0.0.19_INCALL]=[true] 2009-01-21 03:08:09 [DEBUG] switch_core_state_machine.c:416 switch_core_session_run() Hangup Command set_global(10.0.0.19_INCALL=false): It does not execute - the global variable is not set...quite odd. Scott

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Wideband/Ultrawideband Conference
by Brian West 24 Jan '09

24 Jan '09

FreeSWITCHers, We have set the 888(a)conference.freeswitch.org conference to be 32k, This weekend I'm going to move it to be 48k by default. So dust off your Logitech 350 headsets ( or go buy one from your favorite retailer ). The 350 has the 16kHz pickup and the 20kHz output required to take the most advantage of the 48k CELT codec. So check it out... btw we hangout in the conference all day on Fridays so if you want to call in, ask questions or just interact with the developers of FreeSWITCH... Stop by ... say Hi... Thanks, /b PS: Don't forget to tell a friend about FreeSWITCH!

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bgapi uuid_kill question
by pauld 24 Jan '09

24 Jan '09

Hi, Trying to kill a channel (hangup) when it's in CHANNEL_EXECUTE state as per mod_socket_event event information I use bgapi uuid_kill <core_uuid> - but get response "ERR no such channel!". Same happens if I use unique_id or channel_name as an argument to uuid_kill. What I am doing wrong? Help would much appreciated. Thx.

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My fs_cli stopped working, can you help me please?
by Milena 23 Jan '09

23 Jan '09

I use the stable version of fs 1.0.2 testing out different options of configuration; Last thing I was doing was trying to fix some issues with dingaling-google talk cause i hear no audio at all from an external ip, and i still didnt get it to work; so i tried deleting the sofia_* files on the folder db/ as suggested on another thread about nat related issues (oops?), and then restarting fs. After this, trying to execute bin/fs_cli doesn't work anymore; it tries to connect cause i get the output on the box: 2009-01-23 16:23:59 [DEBUG] mod_event_socket.c:1791 listener_run() Connection Open from 127.0.0.1:33055 2009-01-23 16:23:59 [DEBUG] mod_event_socket.c:1914 listener_run() Session complete, waiting for children 2009-01-23 16:23:59 [DEBUG] mod_event_socket.c:1938 listener_run() Connection Closed but on the console where im trying to run the cli nothing happens, I tried rebooting the machine; i tried with freeswitch make current, it didn't fix it; i tried all the way from configure to make and make install and it didn't fix my problem either. What should i run or change on the configurations to fix this? Thank you very much.

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Gateway Params
by Laurent Fabre 23 Jan '09

23 Jan '09

Hi, Can I enclose the params of the directory gateway element in <params></params> ? I mean I sure can but is the parser going to choke on it or not ? Thanks in advance, -- Laurent FABRE Directeur général 10, rue d'Aumale 75009 Paris Tel: +33.(0)1.42.81.28.20<sip:+33142812820> Mob: +33.(0)6.75.75.02.96<sip:+33675750296> Fax: +33.(0)1.70.24.74.61<+33170247461> laurent.fabre(a)kirranet.com

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Few question regarding move from Asterisk to FS - resend
by Ivica Samija 23 Jan '09

23 Jan '09

Last message was incomplete, sorry for that, resending. Hi all, our company have implemented two Asterisk servers to: - connect two company sites - transition to IP telephony - cut down TCO regarding telephony Our interconnection schema: --T1/E1 provider1--< > --T1/E1 provider2--< Asterisk1 >--T1/E1/ trunk--< propriety PBX1 > ---SIP provider3---< > | SIP trunk | < Asterisk2 >--T1/E1/ trunk--< propriety PBX2 > On each site we have number of IP phones connected to Asterisk and analog phones connected to propriety PBX. Features implemented on Asterisk boxes are: - IVR - queue - conference - DISA - BLF - transfer calls All was more or less good until we had to implement call rating (we have to keep track cost made on each extension for statistic). Company policy is that implementation has to be in house. We hit brick wall because Asterisk have inaccurate CDRs (transfers, forwards,...) I am looking in FS for last few weeks and it seams to me that it can replace our Asterisk boxes, and more :). I am a little confused with XML config but it seams to me that it is worth of learning. For most of my question I have found answers in documentation wiki and on list archive, but I have just a few question still without answer. I am sorry if the answers are out there but I was to clumsy to find them. In that case some info or link would be great. So here we go: 1) OpenZap is stable enough that it can be used in production? As you can see we depend on 4 zap trunks. We use OpenVox T1/E1 cards (D210P and D410P) with wct4xxp module. We use ccs, hdb3, crc4, loadzone=it in zaptel.conf We use switchtype=euroisdn, pridialplan=international, pridialplan=unknown, pri_cpe and pri_net signaling. Any suggestions are welcome. 2) Is it possible do bridge to Zap group instead of channel something like Dial(Zap/g4/${EXTEN:1},60,t)? 3) If I understood correct I can talk with SQLite (or other DB) only through Lua, Javascript,... there is nothing similar to Set(mobile=${DB(mob/${EXTEN:1})}) 4) mod_nibblebill, is it possible to get resulting cost per channel in some channel variable and than put value in custom cdr field Sorry on bad english, not a native speaker. Best regards, Ivica

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Few question regarding move from Asterisk to FS
by Ivica Samija 23 Jan '09

23 Jan '09

Hi all, our company have implemented two Asterisk servers to: - connect two company sites - transition to IP telephony - cut down TCO regarding telephony Our interconnection schema: --T1/E1 provider1--< > --T1/E1 provider2--< Asterisk1 >--T1/E1/ trunk--< propriety PBX1 > ---SIP provider3---< > | SIP trunk | < Asterisk2 >--T1/E1/ trunk--< propriety PBX2 > On each site we have number of IP phones connected to Asterisk and analog phones connected to propriety PBX. Features implemented on Asterisk boxes are: - IVR - queue - conference - DISA - BLF - transfer calls All was more or less good until we had to implement call rating (we have to keep track cost made on each extension for statistic). Company policy is that implementation has to be in house. We hit brick wall because Asterisk have inaccurate CDRs (transfers, forwards,...) I am looking in FS for last few weeks and it seams to me that it can replace our Asterisk boxes, and more :). I am a little confused with XML config but it seams to me that it is worth of learning. For most of my question I have found answers in documentation wiki and on list archive, but I have just a few question still without answer. I am sorry if the answers are out there but I was to clumsy to find them. In that case some info or link woud be great. So here we go: 1) OpenZap is stable enough that it can be used in production? As you can see we depend on 4 zap trunks.

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Conference and socket outbound
by Dennis 23 Jan '09

23 Jan '09

hi, now, that everything is working, what i wanted to do with freeswitch, i want to look a little bit more left and right. one interesting thing to "play" arround with, is conferencing. is it possible to do conference calls, when using socket outbound and php? where could i start? ok, of course with "mod conference" in the wiki. but i wonder, how i can put a caller in a conference room and how to set the properties (what dtmf does what and which soundfiles are played when) dynamically. let's say i get a socket inbound call and i answer it. then i play a soundfile, which asks to enter the pin... how can i move the inbound call or originated calls into a conference room? it seems, as if one can do a lot with api in the conference room. but i wonder what i have to do in the beginning, to put people (specific uuid's) into a conference room. thanks, dennis

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voicemail web interface
by Helmut Kuper 23 Jan '09

23 Jan '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I tried to get voicemail web interface up and running. I found that we need xml_rpc to open the 8080 port of FS. Then USer-profile needs "http-allowed-api=voicemail" parameter. After that I left a message for my account and tried to access my voicemails via web (I replaced fs.ip with my IP): http://fs.ip:8080/api/voicemail/web This led to a web page generating out of web-vm.tpl with 0 (zero) voicemails. If I call my voicebox to listen to the messages, there is one new message available. My phone gets a mwi saying there is 1 new message for me. I wonder how the page knows for whom it shall display the voicemails. After that I tried "http://fs.ip:8080/api/domains/this/api/voicemail/web" This delivered: Error 404 Not Found ABYSS Web Server for XML-RPC For C/C++ version Xmlrpc-c 1.14.99 Hm, any ideas ? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkl4m2gACgkQ4tZeNddg3dzIogCgnZybNeIkfsFbyKzSmusgUP5b hc8AoLYYO8Fi5RHKLEk7eXxUfzj+udIW =ofO3 -----END PGP SIGNATURE-----

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freeswitch memory leak issue
by Simon Leck 23 Jan '09

23 Jan '09

Hi All, I am experiencing memory leak issue in FreeSwitch? Anybody knows how this can be resolved, please email me. Thanks in advance to everybody for your kind assistance. Thanks Simon

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