July 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_spandsp wiki error
by Joseph L. Casale 07 Jul '10

07 Jul '10

The wiki states in the Intro: The old mod_fax, mod_t38gateway, and many of the codecs in FreeSWITCH (mod_voipcodecs) have now merged to one module called mod_spandsp which takes advantage of all the DSP features found in the spandsp library including T.38 endpoint and gateway functionality. Yet the "Installation and configuration" states: mod_spandsp is compiled by default with the freeswitch source tree but is still in testing stage to use it just unload mod_voipcodecs and mod_spandsp and load mod_fax remember to change it in modules.conf.xml mod_spandsp require libtiff. Is that an error, should I fix this? If the old mod_fax, mod_t38gateway and ... modules have been renamed and mod_spandsp is in modules.conf.xml there is nothing to do then? Thanks, jlc

2 2

Channel variables from CLI
by Roger Salloum 07 Jul '10

07 Jul '10

Hi, Is there a way to set channel variables in the CLI? I have created a custom sql that uses channel variables to determine routes, and wanted to test it through the CLI. However, I can't find out how to set the channel variables in the CLI so that the lcr module will have that information to use in the sql statement. Thanks,

3 2

Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 49, Issue 28
by Durmuş Ali Öztürk 07 Jul '10

07 Jul '10

2010/7/7 <freeswitch-users-request(a)lists.freeswitch.org>: Hello, Supplying the dial string to the ManagedSession constructor has solved our problem. We can originate the call and stream a media properly. Thanks a lot for your help. Ali Öztürk > ---------- Weitergeleitete Nachricht ---------- > From: Brian West <brian(a)freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Date: Wed, 7 Jul 2010 08:36:55 -0500 > Subject: Re: [Freeswitch-users] Sound not OK (Choppy Sound) while using ManagedSession (.NET/C#) > When a call is parked you can send commands to it to do anything you want. > > > /b > > On Jul 7, 2010, at 8:30 AM, Phillip Jones wrote: > >> In example two you are parking the call. You are then playing media into a parked call. I am not sure this is valid. According to the wiki "Please note that to retrieve a call that has been "parked", you'll have to bridge to them or transfer the call to a valid location." - so you might need to transfer that parked call to a DialPlan App before you try and play media - or just not park it in the first place. >> >> I might be wrong - but that where I would start. > > > > > > ---------- Weitergeleitete Nachricht ---------- > From: Anthony Minessale <anthony.minessale(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Wed, 7 Jul 2010 08:41:03 -0500 > Subject: Re: [Freeswitch-users] Sound not OK (Choppy Sound) while using ManagedSession (.NET/C#) > instead of originate to park then sending the uuid to the constructor > try supplying the dial string to the constructor which will place the call > > On Wed, Jul 7, 2010 at 8:30 AM, Phillip Jones <pjintheusa(a)gmail.com> wrote: >> >> In example two you are parking the call. You are then playing media into a parked call. I am not sure this is valid. According to the wiki "Please note that to retrieve a call that has been "parked", you'll have to bridge to them or transfer the call to a valid location." - so you might need to transfer that parked call to a DialPlan App before you try and play media - or just not park it in the first place. >> >> I might be wrong - but that where I would start. >> >> On Wed, Jul 7, 2010 at 8:03 AM, Durmuş Ali Öztürk <ali.stgt(a)gmail.com> wrote: >>> >>> Hello, >>> >>> I have successfully entegrated a dll-module (written udner .Net / c#) >>> in fs which is loadable by the mod_managed component. >>> >>> Now, if I try to stream a wav file to the callee then the sound is >>> very poor and choppy (also the file is played slower). >>> >>> The fs is running under Windows XP and tested under Windows 2003 with >>> the same result. >>> >>> Streaming by using FreeSWITCH.Native.Api() works without problems, >>> sound is perfect. >>> >>> Samples which I have tried: >>> >>> >>> >>> 1 - This works fine but delegate mechanism is missing there: >>> >>> FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api(); >>> >>> string uuid = fsApi.ExecuteString("create_uuid"); >>> string apiResult = fsApi.Execute("originate", >>> string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/Test/{2} >>> &&playback({3}", uuid, callerID, phoneNumber, wavFile)); >>> >>> >>> >>> >>> 2 - ManagedSession is integrated but we have sound problems: >>> >>> FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api(); >>> >>> string uuid = fsApi.ExecuteString("create_uuid"); >>> >>> string apiResult = fsApi.Execute("originate", >>> string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/Test/{2} >>> &park", uuid, callerID, phoneNumber)); >>> >>> FreeSWITCH.Native.ManagedSession blegSession = new >>> FreeSWITCH.Native.ManagedSession(uuid); >>> >>> if (blegSession.IsAvailable) >>> { >>> while (!blegSession.answered()) >>> { >>> blegSession.sleep(500, 1); >>> } >>> >>> if (blegSession.Ready() && blegSession.mediaReady()) >>> { >>> //blegSession.Answer(); >>> //blegSession.Execute("playback",wavFile); >>> blegSession.StreamFile(wavFile, 0); >>> } >>> >>> blegSession.Hangup("Normal call clearing"); >>> } >>> >>> >>> >>> >>> Is there a mistake in my code or do I have forgotten something? Can >>> you support me with some examples? >>> >>> >>> Another issue is, that I am not able to make a origination directly by >>> ManagedSession or CoreSession without using the >>> FreeSWITCH.Native.Api(). How can I create an instance of a CoreSession >>> object? >>> >>> >>> Many thanks for you help in advance. >>> >>> Ali >>> >>> _______________________________________________ >>> 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 >> >> >> _______________________________________________ >> 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 >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > > ---------- Weitergeleitete Nachricht ---------- > From: Anthony Minessale <anthony.minessale(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Wed, 7 Jul 2010 08:46:08 -0500 > Subject: Re: [Freeswitch-users] Behaviour of group_confirm_cancel_timeout > This is not at all the same. > in your script you have to do one of the following: > if they dialed the right digits answer the channel, if not hangup the channel. > you are doing neither and originate is hanging up for you because you did not answer the channel. > Can we be done with this thread now? it's trying my patience. > > > On Wed, Jul 7, 2010 at 12:47 AM, Nagalenoj H. <nagalenoj(a)gmail.com> wrote: >> >> After git pull, the behavior remains the same. >> >> When I check the console log, the leg has got disconnected after leg_timeout seconds but the session for the leg has got closed after the script exists. >> >> 2010-07-07 05:27:39.156085 [INFO] mod_dialplan_xml.c:331 Processing 1005->212 in context default >> 2010-07-07 05:27:39.194359 [NOTICE] mod_dptools.c:746 Channel [sofia/internal/1005(a)192.168.1.72] has been answered >> 2010-07-07 05:27:42.708162 [NOTICE] sofia.c:4875 Channel [sofia/internal/sip:1000@192.168.6.114] has been answered >> 2010-07-07 05:27:49.020053 [INFO] mod_dptools.c:2393 Originate Failed. Cause: NO_ANSWER >> 2010-07-07 05:28:12.761902 [NOTICE] switch_core_session.c:1193 Session 6 (sofia/internal/sip:1000@192.168.6.114) Ended >> 2010-07-07 05:28:12.761902 [NOTICE] switch_core_session.c:1195 Close Channel sofia/internal/sip:1000@192.168.6.114 [CS_DESTROY] >> >> Here is the console log, >> http://pastebin.freeswitch.org/13395 >> >> On Tue, Jul 6, 2010 at 8:56 PM, Anthony Minessale <anthony.minessale(a)gmail.com> wrote: >>> >>> try latest GIT, that was an edge case and it's fixed. >>> >>> On Tue, Jul 6, 2010 at 1:58 AM, Nagalenoj H. <nagalenoj(a)gmail.com> wrote: >>>> >>>> Got core dump when I execute the bridge as >>>> execute-app-arg: {user_recurse_variables=false,group_confirm_cancel_timeout=true}[leg_timeout=10]user/1000 >>>> >>>> Console log is here, >>>> http://pastebin.freeswitch.org/13371 >>>> >>>> core dump backtrace is here, >>>> http://pastebin.freeswitch.org/13372 >>>> >>>> Also, tried after git pull, but the result is same. >>>> >>>> On Mon, Jul 5, 2010 at 8:53 PM, Anthony Minessale <anthony.minessale(a)gmail.com> wrote: >>>>> >>>>> you also need user_recurse_variables=false >>>>> {user_recurse_variables=false,group_confirm_cancel_timeout=true} >>>>> >>>>> >>>>> On Sat, Jul 3, 2010 at 8:22 AM, Nagalenoj H. <nagalenoj(a)gmail.com> wrote: >>>>>> >>>>>> Here is the log got from the latest GIT source. >>>>>> >>>>>> http://pastebin.freeswitch.org/13347 >>>>>> >>>>>> On Fri, Jul 2, 2010 at 10:15 PM, Anthony Minessale <anthony.minessale(a)gmail.com> wrote: >>>>>>> >>>>>>> update and reproduce that same log with latest GIT the version you are using has an issue. >>>>>>> >>>>>>> On Fri, Jul 2, 2010 at 12:25 AM, Nagalenoj H. <nagalenoj(a)gmail.com> wrote: >>>>>>>> >>>>>>>> I've pasted the console log here, >>>>>>>> >>>>>>>> http://pastebin.freeswitch.org/13333 >>>>>>>> >>>>>>>> >>>>>>>> On Wed, Jun 30, 2010 at 11:15 PM, Michael Collins <msc(a)freeswitch.org> wrote: >>>>>>>>> >>>>>>>>> Can you supply a console log of these calls? >>>>>>>>> -MC >>>>>>>>> >>>>>>>>> On Wed, Jun 30, 2010 at 7:37 AM, Nagalenoj H. <nagalenoj(a)gmail.com> wrote: >>>>>>>>>> >>>>>>>>>> Dear Anthony, >>>>>>>>>> I've tried using the group_confirm_cancel_timeout as per the discussion we had in IRC. You wanted to used it as part of dial string and not as a channel variable. >>>>>>>>>> But, It doesn't work for me. >>>>>>>>>> >>>>>>>>>> Here is how I've given the commands and the script I've executed. Even when I give group_confirm_cancel_timeout, the callee's leg is getting disconnected after legtimeout. >>>>>>>>>> >>>>>>>>>> connect >>>>>>>>>> >>>>>>>>>> sendmsg >>>>>>>>>> call-command: execute >>>>>>>>>> execute-app-name:answer >>>>>>>>>> >>>>>>>>>> sendmsg >>>>>>>>>> call-command: execute >>>>>>>>>> execute-app-name: set >>>>>>>>>> execute-app-arg: group_confirm_key=exec >>>>>>>>>> >>>>>>>>>> sendmsg >>>>>>>>>> call-command: execute >>>>>>>>>> execute-app-name: set >>>>>>>>>> execute-app-arg: group_confirm_file=perl /root/bridge.pl >>>>>>>>>> >>>>>>>>>> sendmsg >>>>>>>>>> call-command: execute >>>>>>>>>> execute-app-name: bridge >>>>>>>>>> execute-app-arg: {group_confirm_cancel_timeout=1}[leg_timeout=10]user/1005 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> bridge.pl: >>>>>>>>>> #!/usr/bin/perl >>>>>>>>>> use freeswitch; >>>>>>>>>> >>>>>>>>>> our $session; >>>>>>>>>> freeswitch::consoleLog("info","Goint to get the digits"); >>>>>>>>>> # To simulate the scenario I used sleep here. >>>>>>>>>> sleep(30); >>>>>>>>>> 1; >>>>>>>>>> >>>>>>>>>> Kindly tell me whats wrong in the above. >>>>>>>>> >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> 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 >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Regards, >>>>>>>> Nagalenoj H. >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> 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 >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Anthony Minessale II >>>>>>> >>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>> ClueCon http://www.cluecon.com/ >>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>>>> >>>>>>> AIM: anthm >>>>>>> MSN:anthony_minessale@hotmail.com >>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >>>>>>> IRC: irc.freenode.net #freeswitch >>>>>>> >>>>>>> FreeSWITCH Developer Conference >>>>>>> sip:888@conference.freeswitch.org >>>>>>> googletalk:conf+888@conference.freeswitch.org >>>>>>> pstn:+19193869900 >>>>>>> >>>>>>> _______________________________________________ >>>>>>> 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 >>>>>>> >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> Nagalenoj H. >>>>>> >>>>>> _______________________________________________ >>>>>> 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 >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale@hotmail.com >>>>> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >>>>> IRC: irc.freenode.net #freeswitch >>>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888@conference.freeswitch.org >>>>> googletalk:conf+888@conference.freeswitch.org >>>>> pstn:+19193869900 >>>>> >>>>> _______________________________________________ >>>>> 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 >>>>> >>>> >>>> -- >>>> Regards, >>>> Nagalenoj H. >>>> >>>> _______________________________________________ >>>> 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 >>>> >>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale@hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888@conference.freeswitch.org >>> googletalk:conf+888@conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _______________________________________________ >>> 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 >>> >> >> -- >> Regards, >> Nagalenoj H. >> >> _______________________________________________ >> 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 >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > 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 > >

1 0

G729 validator not working on Debian 4.0
by David Ponzone 07 Jul '10

07 Jul '10

Hi Everyone, I am trying to install the G729 codec on a Debian 4.0. On 2 different boxes, validator fails with: Floating point exception One box is a 64bit Debian 4.0 running on Intel(R) Core(TM)2 Duo CPU E6550 @ 2.33GHz. The other one is a 32 bit Debian 4.0 running on dual Intel(R) Xeon(TM) CPU 2.80GHz. Both boxes have been installed by totally different people, so there is no chance that they had been doing some identical strange things during install. Is there a known issue with Etch ? Do I have any chance to fix it ? Thank you David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

2 2

Sound not OK (Choppy Sound) while using ManagedSession (.NET/C#)
by Durmuş Ali Öztürk 07 Jul '10

07 Jul '10

Hello, I have successfully entegrated a dll-module (written udner .Net / c#) in fs which is loadable by the mod_managed component. Now, if I try to stream a wav file to the callee then the sound is very poor and choppy (also the file is played slower). The fs is running under Windows XP and tested under Windows 2003 with the same result. Streaming by using FreeSWITCH.Native.Api() works without problems, sound is perfect. Samples which I have tried: 1 - This works fine but delegate mechanism is missing there: FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api(); string uuid = fsApi.ExecuteString("create_uuid"); string apiResult = fsApi.Execute("originate", string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/Test/{2} &&playback({3}", uuid, callerID, phoneNumber, wavFile)); 2 - ManagedSession is integrated but we have sound problems: FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api(); string uuid = fsApi.ExecuteString("create_uuid"); string apiResult = fsApi.Execute("originate", string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/Test/{2} &park", uuid, callerID, phoneNumber)); FreeSWITCH.Native.ManagedSession blegSession = new FreeSWITCH.Native.ManagedSession(uuid); if (blegSession.IsAvailable) { while (!blegSession.answered()) { blegSession.sleep(500, 1); } if (blegSession.Ready() && blegSession.mediaReady()) { //blegSession.Answer(); //blegSession.Execute("playback",wavFile); blegSession.StreamFile(wavFile, 0); } blegSession.Hangup("Normal call clearing"); } Is there a mistake in my code or do I have forgotten something? Can you support me with some examples? Another issue is, that I am not able to make a origination directly by ManagedSession or CoreSession without using the FreeSWITCH.Native.Api(). How can I create an instance of a CoreSession object? Many thanks for you help in advance. Ali

4 3

Only Eutelia not working on NAT
by xuyan yang 06 Jul '10

06 Jul '10

My freeswitch is working fine behind a NAT with 12voip, sipgate, voipcheap, VoXalot as my router support UPnP. Recently, I have applied some DID numbers from Eutelia.it and tested the account with x-lite. But when configured to FS, it can only make out-going call and fail to receive any form this gateway. In the meanwhile 12voip,sipgate are still working well. I tried many parameters but none of them works unless the router was removed and FS connected directly to public IP. It is quite strange for only a single gateway fail behind NAT, while all others is fine.

3 3

408 error?
by Lars Zeb 06 Jul '10

06 Jul '10

I am getting a 408 error. I do not understand why. I am using git-1fba654 2010-07-01 on Centos 5.3. Could it be Flowroute? Thanks for any help, Lars http://pastebin.freeswitch.org/13389

2 2

bind_meta_app modification
by Bradley Brashier 06 Jul '10

06 Jul '10

Our design needed something like bind_meta_app, but couldn't use the * key. So we made a modification to allow you to use different keys via an xml variable. I assume that this would also be useful to others. The specific change is in switch_ivr_async.c, but my understanding is that bind_meta_app is a dialplan tool... in which section should I submit this to jira? Bradley Brashier Virtual PBX

2 1

IPComms Provider Template
by broken dash 06 Jul '10

06 Jul '10

I have been trying to setup a sip gateway using fusionpbx but for whatever reason the web gui isnt working 100% correctly.Does anyone have a template config for IPComms so I can manually include the XML into freeswitch? Thanks, Brian

1 0

Dynamic mod_conference configuration
by Tim St. Pierre 06 Jul '10

06 Jul '10

Hello list, I am trying to set up our conferencing system to pull some preferences from a database. In the conference.conf.xml file, I have a few profiles that cover pretty much all the bases, but I want to be able to set the music on hold sound differently for each conference, without reloading XML. I tried using ${hold_music} as the moh_sound in the config file, but this doesn't seem to expand. Is there a way to dynamically set conference profile options with a channel variable? I read a bit on generating config with LUA. The wiki talks about dialplan and directory, but doesn't really say much about other module configs. Can I bind a lua script to mod_conference? Can someone give me a snippet of that binding if that's possible? Thanks for the help! -Tim

2 1