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mod_h323 load error - undefined symbol
by Steven Ayre 02 Nov '10

02 Nov '10

I'm having trouble loading mod_h323. This is the error I get when I try loading the module: freeswitch@ss3> load mod_h323 2010-11-02 10:16:55.175574 [CRIT] switch_loadable_module.c:928 Error Loading module /opt/freeswitch/mod/mod_h323.so **/opt/freeswitch/mod/mod_h323.so: undefined symbol: _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi** I have ptlib2.8.2 installed and h323plus trunk as per the instructions on the Wiki. The problem appears to be that mod_h323 expects _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi That isn't in libh323.so, but _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infol is (ends L not I): ss3:/usr/lib# strings libh323.so | grep OnUserInputInlineRFC2833 _ZNK14H323Connection34OnUserInputInlineRFC2833_PNotifier4CallER7PObjectl _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infol <--- closest match! _ZN14H323Connection34OnUserInputInlineRFC2833_PNotifierD1Ev _ZN14H323Connection34OnUserInputInlineRFC2833_PNotifierD0Ev _ZTVN14H323Connection34OnUserInputInlineRFC2833_PNotifierE _ZTIN14H323Connection34OnUserInputInlineRFC2833_PNotifierE _ZTSN14H323Connection34OnUserInputInlineRFC2833_PNotifierE N14H323Connection34OnUserInputInlineRFC2833_PNotifierE Does anyone know what'd cause this and how to fix it? Platform is 64bit Debian Lenny. -Steve

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Any info on mod_sndfile? what is it used for? It is not documented although it is large !
by Majdi Bsoul 02 Nov '10

02 Nov '10

Hello guys, I am trying to minimize the footprint of FreeSWITCH runtime and remove un-needed modules and their libs. This modules is not mentioned any where and not much on the web. Could u please provide more info on what functionalities it provides? I'm planning to use FreeSWITCH for only SIP/sofia and Jingle/gtalk. Thanks,

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FXO port manual off-hook
by Jeroen C. van Gelderen 02 Nov '10

02 Nov '10

Hi Guys, I don't think this is currently supported but... Can I force (and keep) an FXO (FreeTDM/DAHDI) port off-hook from the dialplan or from a script? Right now when the CO sends a CPC (disconnect supervision) signal the receiving FXO port automatically goes on-hook. I'd like to be able to "busy" a line (take it off-hook and keep it off-hook) so that calls will roll-over to other lines in my CO hunt group. If not supported, could this be implemented in a relatively sane amount of time by a non-Freeswitch expert programmer? Or would this require major re-architecting? Any and all pointers appreciated. Cheers, -Slim -- Jeroen C. "Slim" van Gelderen

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ivr terminitor key?
by Seven Du 01 Nov '10

01 Nov '10

Hi, For ivr menus that have digit-len > 1, is there a way to specify a terminator key, say #, to stop digits collecting immediately? Default behavior tends to wait until a timeout. Thanks. -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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Probably obvious but not to me - sofia hangup
by general＠bestoronto.com 01 Nov '10

01 Nov '10

Hi, I am on freeswitch / centOS: Linux pbx 2.6.18-194.11.3.el5.centos.plus #1 SMP It is a fairly straight out of the box freeswitch config, I am not so good with XML yet. Issue is the line to my single SIP phone will drop after about 8 minutes, no matter if talking or not. What might be the cause of this? Seems like a timeout or lack of ACK somewhere? Thanks for the help.... Mark B

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How to generate ringing when dialing out through dingaling/gtalk
by Majdi Bsoul 01 Nov '10

01 Nov '10

How can I generate a ringing tone treatment to the dialing extension, when dialing through dingaling/gtalk? Currently the extension user gets dead air, until the destination answers. I am running with the most recent GIT FreeSWITCH (10/29/2010). Regards,

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conference profile question
by covici＠ccs.covici.com 31 Oct '10

31 Oct '10

Hi. I have some conferences setup, not using the default profile. Now if the conference calls someone using the dial command, he seems to come in with the default profile -- any variable or other thing I can set so he comes in with the profile I called the conference originally with by the conference app, or a better way to accomplish this? thanks. -- 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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Problem with bind_meta_app
by Mike van Lammeren 30 Oct '10

30 Oct '10

Hello! I want *1 to trigger a Lua script, so I configured my dialplan with a call to bind_meta_app like this: <extension name="my_bridge_call"> <condition field="destination_number" expression="^11"> <action application="bind_meta_app" data="1 b b lua::my_trigger.lua"/> <action application="lua" data="my_bridge_call.lua"/> </condition> </extension> With the configuration above, whenever I press *1 on the b leg, the my_trigger.lua script is launched. That part works great! However, it seems like something else is also listening for the *1, since notice a few oddities every time I trigger the script: 1. I see this error in the console: 2010-11-05 11:38:53.875766 [ERR] switch_core_file.c:122 Invalid file format [local_stream] for [moh]! 2. On the a leg, I can hear a noise when dialing *1 on the b leg. I don't quite hear the touch tones, but the noise is definitely associated with them. 3. After triggering the script, no audio is transferred for about 5 seconds. It looks to me like FreeSWITCH is trying to transfer the calls to music-on-hold, then gets an error, times out, and goes back to the original call. There are no other calls to bind_meta_app, so what else is reacting to the *1? Thanks! Mike van Lammeren

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2010-11-07 17:17:49.876400 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_dingaling.so , **/usr/local/freeswitch/mod/mod_dingaling.so: undefined symbol: switch_channel_get_variable** , 2
by Pekka Kurki 30 Oct '10

30 Oct '10

get with the latest snapshot load error: 2010-11-07 17:17:49.876400 [CRIT] switch_loadable_module.c:926 Error Loading module /usr/local/freeswitch/mod/mod_dingaling.so **/usr/local/freeswitch/mod/mod_dingaling.so: undefined symbol: switch_channel_get_variable** what is wrong? br ---pekka---

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Loss of first second of media
by Fraser Redmond 30 Oct '10

30 Oct '10

I've been trying out a few new termination providers, and am finding that the first second of media is being lost. I get early media (ringing), then when the call is answered the first half a second to a second is lost, then the rest of the call is fine. The engineers at one of the providers ran a sip trace and a full packet trace, and it looked like everything was being delivered from the pstn to my server. They said that none of their other customers have reported this problem before. I've tested connecting to pstn's in several different countries, with the same results. I've used a couple of other termination providers in the past (1 still in current use) which have been fine, but now have 2 that are losing the first part of the media. So it seems like it's specific to Freeswitch, with some subset of conditions for some providers. This is running on Ubuntu on a Amazon AWS server, in case that helps. Any ideas how to narrow it down further? Cheers, Fraser

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