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Can FreeSWITCH act as a turn server for ICE?
by Fabio Pietrosanti (naif) 02 Oct '10

02 Oct '10

I was reading an interesting approach that SER had taken in making the mediaproxy (rtp proxy) component be announced as a SDP candidate: http://mediaproxy.ag-projects.com/wiki/ICE That way the SIP client has to talk ICE and STUN but without the requirement to run TURN (that's very few supported). Would be already possible somehow to have a setup like this with FreeSWITCH having it working also as the role of a TURN server but without even implementing TURN? Fabio

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mod skype legal
by budi wibowo 02 Oct '10

02 Oct '10

hi it's nice to know that FS have module to integrate with skype. is it something that skype supported and legally allowed? say, i have project on mapping some phone number to some skype user. is it safe to use the module for commercial use? regards budi wibowo

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distributor in LCR
by Octavio Duarte 02 Oct '10

02 Oct '10

Hello Here is my problem, i have a several gateways and each one of them gw has a different calls' capacity but the same rate , what i wanna do is to use something like mod distributor to be able to distribute call among gw´s according it's capacity, i dont want to full each gateway to send calls to another empty. In LCR module i read that i can use random to get different order, but that is not what i need! i see that i can add custom sql but i wanted to know if there is something already done! Finally I dont see the way to use mod limit and distributor together i am really new on FS but i seems to be an amazing project! Thank in advance for you help and time! Best Regards from Mexico

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Openzap rxgain and txgain not working well
by sdemers＠q4innovation.com 02 Oct '10

02 Oct '10

Hi everyone, I notice that the rxgain and txgain in the openzap.conf is not working for the first span (span 1) Is it an issue? Here is the result of my investigation. Configuration: Latest Freeswitch (September 9, 2010) WANPIPE Release: 3.5.15 without smg_pri Sangoma A108 with Echo cancellation enabled. Wanpipe and libpri. In the following openzap.conf, the telus2 got the gains but not telus1 The logs indicate that both got the configuration correctly... But the recording session demonstrate that they don't... [span wanpipe telus1] trunk_type => T1 b-channel => 1:1-23 d-channel => 1:24 rxgain => 11.0 txgain => 7.0 [span wanpipe telus2] trunk_type => T1 b-channel => 2:1-23 d-channel => 2:24 rxgain => 11.0 txgain => 7.0 WORKAROUND: Then I maid a dummy span at the top of these two configs... Yes the dummy span doesn't exists, in fact there is no T1 connected to it. I got some opening error, but I got the gains on both span now... ex: [span wanpipe dummy] trunk_type => T1 b-channel => 3:1-23 d-channel => 3:24 rxgain => 11.0 txgain => 7.0 [span wanpipe telus1] trunk_type => T1 b-channel => 1:1-23 d-channel => 1:24 rxgain => 11.0 txgain => 7.0 [span wanpipe telus2] trunk_type => T1 b-channel => 2:1-23 d-channel => 2:24 rxgain => 11.0 txgain => 7.0 ------------------------------- Openzap.xml.conf Libpri configuration <libpri_spans> <span name="telus1"> <param name="q921loglevel" value="alert"/> <param name="q931loglevel" value="alert"/> <param name="node" value="user"/> <param name="switch" value="ni2"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> </span> <span name="telus2"> <param name="q921loglevel" value="alert"/> <param name="q931loglevel" value="alert"/> <param name="node" value="user"/> <param name="switch" value="ni2"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> </span> </libpri_spans>

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Error compiling mod_callcenter
by Rob Hutton 02 Oct '10

02 Oct '10

When trying to compile mod_callcenter I am getting the following error: making all mod_callcenter Compiling /usr/src/freeswitch/freeswitch/src/mod/applications/mod_callcenter/mod_callcenter.c... quiet_libtool: compile: gcc -I/usr/src/freeswitch/freeswitch/src/include - I/usr/src/freeswitch/freeswitch/src/include - I/usr/src/freeswitch/freeswitch/libs/libteletone/src -fPIC -Werror - fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -g - O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE - DHAVE_CONFIG_H -c /usr/src/freeswitch/freeswitch/src/mod/applications/mod_callcenter/mod_callcenter.c -fPIC -DPIC -o .libs/mod_callcenter.o cc1: warnings being treated as errors /usr/src/freeswitch/freeswitch/src/mod/applications/mod_callcenter/mod_callcenter.c: In function ‘cc_config_api_function’: /usr/src/freeswitch/freeswitch/src/mod/applications/mod_callcenter/mod_callcenter.c:2466:6: error: case value ‘11’ not in enumerated type ‘cc_agent_status_t’ make[5]: *** [mod_callcenter.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_callcenter-all] Error 1 make[2]: *** [all-recursive] Error 1 According to cc_agent_status_t; 0, 1, 2, 3, and 4 are valid. The switch statement this refers to tests for CC_STATUS_INVALID_KEY which is defined in cc_status_t as the numeric value 11. The switch statement calls cc_agent_get which defines the return value as cc_agent_status_t. In looking at that function, it seems that the return values referenced all come from cc_status_t, so I think just changing the defined returned value fixes the issue, but there are some select statements that I am not sure what the possible values are. -- Thanks, Rob

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Multiple "voices" under mod_say_en
by Neil Burgess 02 Oct '10

02 Oct '10

Hi, I'd like to be able to use two sets of language files for US and UK within the say module, allowing either to be utilised with the spell, items, etc capabilities, based on setting some channel variable! At the moment, I can't see how to achieve this since it appears that the "say_en" module for manipulating these strings is linked to the language name attribute as below. <language name="en" sound-path="$${base_dir}/sounds/en/us/callie" tts-engine="unimrcp:loquendo7-mrcp2" tts-voice="kate"> Is there any way of supporting multiple "voices" under the same mod_say_en Thanks, Neil

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user reachability
by babak yakhchali 02 Oct '10

02 Oct '10

Hi Is there anything like nat-options-ping and unregister-on-options-fail to use for all registered users (not just users behind nat)? cause sometimes the network connection between users and fs breaks and I don't want to send calls to those users. thanx

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voicemail email not working.
by covici＠ccs.covici.com 01 Oct '10

01 Oct '10

Hi. I left a test message and instead of using the variable I have in my user directory, it tried to deliver to username@domain name which is invalid -- so why did it not use the variable voicemail_email I have in the user directory and it completely ignored the voicemail_notify_email I have as well? Any assistance would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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Simultaneous ring of all phones registered with the same extension number
by Vitalii Colosov 01 Oct '10

01 Oct '10

Hi, Could not find this answer, so would appreciate any hints. I have registered 2 sip phones using the *same *extension number. When I dial this extension number, only one sip phone rings. However I need both of them to start ringing. Is it possible, that when I dial the number - both other phones will ring, not only one..? And when I answer on a phone, another one would stop ringing. Thank you, Vitalie

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Echo on sip trunk
by Alan Dawson 01 Oct '10

01 Oct '10

Hi, I have a freeswitch box, with a sip trunk to a voip provider. When making / receiving external calls the external party gets an echo, which the freeswitch users dont suffer from. What can I do to reduce that. Thanks Alan Dawson -- GPG key: http://aland.burngreave.net/files/e81a4bba.gpg.pub.asc Key Transition: http://aland.burngreave.net/files/keytransition.txt.asc Further Reading: https://we.riseup.net/alster/openpgp-dsa1-key-rollover

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