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ITSPA 2012 Award for Open Source VoIP Projects
by Daniel-Constantin Mierla 23 Mar '12

23 Mar '12

Hello, ITSPA UK has unveiled the winners of its 4th annual Awards, an event designed to celebrate innovation and best practice in the VoIP industry: * http://www.itspaawards.org.uk/ Open Source VoIP Projects won a special category this year, Members' Pick, for providing a real value to VoIP Industry. I had the chance to attend the event in London and I have been selected to pick up the award. I made a news on the website of the project I am mainly involved in (Kamailio) with more details: * http://www.kamailio.org/w/2012/03/itspa-awards-2012-open-source-voip-projec… As you would expect, a complete voip platform usually involves several open source projects, for components such as load balancers, registrar, proxy, gateways or media servers, thus the decision of ITSPA for awarding to the group. FreeSwitch was a frequent presence in the VoIP systems of the ITSPA members I spoke to. I guess it is no surprise at all, you know it very well from the community. Therefore the award happened due to a significant contribution of FreeSwitch to IP Telephony industry as well. As another user of FreeSwitch project, I take the opportunity to thank again to the people behind the project. This time, special wishes to Brian West - fast and smooth recovery, come back quickly to full "ultra wide band" capacity! If anyone is looking for more insights (for news, blogs, personal curiosity) about the event, just drop me an email! Cheers, Daniel -- Daniel-Constantin Mierla Co-Founder Kamailio SIP Server - http://www.kamailio.org Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

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Re: [Freeswitch-users] Internal VM Audio is below acceptable limit
by Peter Olsson 23 Mar '12

23 Mar '12

Ok - that's the problem then :) If you want to make sure everything "just works" - install on real hardware. If you use the built in tools timer_test and time_test I'm pretty sure it will show you quite bad timing. It's not impossible to succeed with a virtual setup, but it will require lots of testing, and usually you need to be in control of the physical host, so it's possible to tweak some performance values for the virtual machine. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Robert Longfield Skickat: den 22 mars 2012 22:06 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Internal VM Audio is below acceptable limit Ah, FS is running on a VPS. From: Brian Foster<mailto:bdfoster@endigotech.com> Sent: Thursday, March 22, 2012 4:04 PM To: FreeSWITCH Users Help<mailto:freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Internal VM Audio is below acceptable limit Actually, I believe what he was asking about was the actual server, although the info you provided already does give us some insight into your setup. Is the freeswitch instance running on virtual or real hardware? -BDF On Mar 22, 2012 3:41 PM, "Robert Longfield" <robert.longfield(a)klinsight.com<mailto:robert.longfield@klinsight.com>> wrote: Hey Peter, Thank for the question. We have a mixture of softphones on computers and mobiles and handsets connected on a ATA box. It seems that the handsets can leave VM but the softphones cannot. I would never have thought of grabbing a handset to test this. For the office this won't be a problem for too long as we are moving to phones but we do have the option of working from home and those cases require softphones to work. -Rob -----Original Message----- From: Peter Olsson Sent: Thursday, March 22, 2012 3:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Internal VM Audio is below acceptable limit Are you on real hardware, or on a virtual machine? /Peter ________________________________________ Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] för Robert Longfield [robert.longfield(a)klinsight.com<mailto:robert.longfield@klinsight.com>] Skickat: den 22 mars 2012 20:16 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] Internal VM Audio is below acceptable limit I have FS up and running with the awesome help of this community. I have a few minor bugs to workout. Some of my own doing (music on hold) but one that I can't trackdown or figure out the cause is when someone internally calls an extension and tries to leave a voice mail the system says something to the effect that the audio was below acceptable limit, basically it was too short. If the person at the extension picks up there is a slight pause and then you can talk with no issues and the audio sounds great. If the same extension is called from an outside line and the call goes to VM there are no problems leaving a message. Pastebin of the log http://pastebin.freeswitch.org/18723 -Rob _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f6b937d32761551985203!

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Need help on freeswitch conference
by haloha 23 Mar '12

23 Mar '12

Hi List i have problem with freeswitch conference example: 1. a company A with PSTN phone number 0822234562 - auto answer machine(IVR) and an extension of company A is from 1000 - 1990 2. a normal PSTN phone : 0833334512 i setup a freeswitch conference room with number is 50000, a moderator dial *2 to make out bound call my network topology: extension A and extension B ------> FS ------SP ------PSTN extension A is a moderator of conference room 50000 entension A calls 50000 and conference is created successfull so there are 2 cases : 1. extension A dial *2 and then enter 0833334512, 0833334512 is added to conference 50000 successfull and everything is fine 2 extension A dial *2 and enter 0822234562, 0822234562 is added to conference successfull The problem is when extension A dial 0822234562, the IVR is played which ask extension A to enter company A's extension, and extension A dials 1000 but it doesnt work because extension A, IVR are in the conference i do debug and see extension A, IVR in a conference and when extension A dial 1000 and see the DTMF event 1 0 0 0 in a conference so that to make it works i do the command line: conference <confname> dtmf <[member_id|all|last]> <digits> conference 50000 dtmf all 1000 is there a way in a case 2: to let extension A dials 1000 when IVR playing to reach company A's extension 1000 and then add extension A, 1000 into a conference 50000 Thank you Ha`

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Re: [Freeswitch-users] mod h323 error
by Peter Olsson 23 Mar '12

23 Mar '12

It seems you might need to modify the Makefile, because of the errors reported. Please read through this information first: http://wiki.freeswitch.org/wiki/Mod_h323 I personally recommend not to use h323plus-trunk. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För budi wibowo Skickat: den 22 mars 2012 13:44 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] mod h323 error hi i just tried to build mod h323 with ptlib 2.8.2 and h323plus-trunk, FreeSWITCH Version 1.0.head (git-9d3401e 2012-03-19 20-06-36 -0500) when i make mod_323 i got this error usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:623: error: 'OPAL_G7231_5k3' was not declared in this scope /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h: At global scope: /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:623: error: 'OPAL_G7231_5k3' has not been declared /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:623: error: expected ',' or '...' before string constant /usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:624: error: expected constructor, destructor, or type conversion before 'class' make[4]: *** [mod_h323.lo] Error 1 make[3]: *** [all] Error 1 make[2]: *** [mod_h323-all] Error 1 make[1]: *** [mod_h323] Error 2 make: *** [mod_h323] Error 2 detail error log http://pastebin.freeswitch.org/18717 help to solve this issue is welcome and appreciated regards budi wibowo !DSPAM:4f6b1d7932768918913259!

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Re: [Freeswitch-users] Originate using inbound socket connection
by Peter Olsson 23 Mar '12

23 Mar '12

If session:ready() becomes false, can't you just exit the script (as you should if it becomes false), and the bridge will be completed? /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Michael Lutz Skickat: den 23 mars 2012 11:47 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Originate using inbound socket connection Thanks used the &park() and it worked!, except.... It seemed the bridge did not take place while my aleg is inside the lua script. I had to finish the script and the the calls where bridged (even the bridge command was allready send), the weird thing is that while I am still inside the lua script after the uuid_bridge, mys session:ready() immedeately becomes [false] while I am still inside the script. The bridge is now initiated on the non-lua side (ESL Connection). Would be an idea to move that to the Luascript using a session:bridge() however I have no session object of the bleg ... Only way to go there is calling the api to do a uuid_bridge as I know the uuid of the bleg, but by using the uuid_bridge in Lua instead of the session:bridge() would problably lead to the same.... Mike. 2012/3/22 Hector Geraldino <Hector.Geraldino(a)ipsoft.com>: > So the problem seems to be with the parameters you're using for the > uuid_bridge command, not your implementation :) > > Can you do a '>show channels' from the fs cli and see what are the uuid's of the active channels in FS? Then use those uuids to perform the uuid_bridge and see what happens. > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of > Michael Lutz > Sent: Wednesday, March 21, 2012 6:05 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Originate using inbound socket > connection > > Nope, it's the same, it only says OK, but nothing else... > > > 2012/3/21 Hector Geraldino <Hector.Geraldino(a)ipsoft.com>: >> What happens if you try to do the same operation from the fs_cli ? >> >> uuid_bridge legA_uuid legB_uuid >> >> Works on the console and not when you call it from your app? >> >> >> -----Original Message----- >> From: freeswitch-users-bounces(a)lists.freeswitch.org >> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of >> Michael Lutz >> Sent: Wednesday, March 21, 2012 5:20 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Originate using inbound socket >> connection >> >> Thanks, that worked. >> >> Now my next problem occurs... ;-) >> >> I am trying to bridge these calls to another allready established >> session in the switch (in a Lua script). >> >> I can reach all data of the newly created session using my created >> uuid which I specified in the originate. >> >> However, the uuid_bridge does not seem to work. (it says +OK and the >> the uuid of the bleg) but everything stays silent and no message is >> being wrtitten to console. >> >> When I check console, I do notice that the new session seems to use a >> different uuid than I specified, (which seems strange as i can access >> al sesssion data uusing uuid_ commands...) >> >> string cDialString = "{origination_uuid=" + legb_uuid + >> ",origination_caller_id_number=" + thisAni + >> ",origination_caller_id_name=" + thisAniName + "}sofia/external/" + >> thisDestination + " &park()"; >> >> var eslEvent = eslDial.Api("originate", cDialString); dispo = >> eslDial.Api("uuid_getvar", legb_uuid + " >> endpoint_disposition").GetBody(); >> while (eslDial.Api("uuid_exists", thisFrom).GetBody() == "true" && >> eslDial.Api("uuid_exists", legb_uuid).GetBody() == "true" && dispo != >> "ANSWER") >> { >> .. >> blabla >> } >> >> when I am connected I just do a "uuid_bridge <the other uuid> legb_uuid" >> >> I get a response: >> +OK <legb_uuid> >> >> My console says: >> New Channel >> sofia/external/316.......(a)xxx.xxx.xxx.xxx[7cb13e5d-1dfe-483e-a8da-ec3 >> a592700c7] >> >> and is a completely different uuid than my legb uuid i specified and >> use to get to the data...... >> >> Any ideas? >> >> Thanks, >> Mike >> >> 2012/3/21 Peter Olsson <peter.olsson(a)visionutveckling.se>: >>> You need to put the other end of the call somewhere >>> >>> The correct string is (example) "originate sofia/gateway/test/1002 &park()" >>> >>> This will call 1002 and then park the call. >>> >>> /Peter >>> >>> >>> -----Ursprungligt meddelande----- >>> Från: freeswitch-users-bounces(a)lists.freeswitch.org >>> [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Michael >>> Lutz >>> Skickat: den 21 mars 2012 14:49 >>> Till: FreeSWITCH Users Help >>> Ämne: [Freeswitch-users] Originate using inbound socket connection >>> >>> Hi Guys, >>> >>> I am working on a .net ESL host (inbound mode) and connect to the ESL using ESLConnection. >>> Now I am trying to originate a call using the ESLConnection.Api() function. But it does't work as expected. >>> >>> It always returns: >>> === >>> Content-Type: api/response >>> Content-Length: 125 >>> Content-Length: 125 >>> >>> -USAGE: <call url> <exten>|&<application_name>(<app_args>) >>> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] >>> === >>> >>> But I am pretty sure my string is right: >>> >>> {origination_caller_id_number=31341.....,origination_caller_id_name= >>> 31341......}sofia/external/31634258...(a)xxx.xxx.xxx.xxx >>> >>> I also tried with double {{ and }} but same result. Even if I take out the whole {} string and just use "sofia/external/etc.." I get the same message back. >>> >>> My code: >>> >>> ESLconnection eslDial = new ESLconnection("x.x.x.x", "8021", "x"); >>> if (eslDial.Connected() == ESL_SUCCESS) { >>> // Create a uuid used to identify the b-leg. >>> string legb_uuid = eslDial.Api("create_uuid", "").GetBody(); >>> string cDialString = "{{origination_uuid=" + legb_uuid + >>> ",origination_caller_id_number=" + thisAni + >>> ",origination_caller_id_name=" + thisAniName + "}}sofia/external/" + >>> thisDestination; >>> >>> // Send the command >>> var eslEvent = eslDial.Api("originate", cDialString); >>> >>> // Write the result to the console >>> Console.WriteLine(eslEvent.Serialize(string.Empty)); >>> return true; >>> } >>> >>> Thanks for your help!, >>> >>> Regards, >>> Michael Lutz >>> >>> ____________________________________________________________________ >>> _____ Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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-u >>> sers >>> http://www.freeswitch.org >>> >>> >>> >>> >>> ____________________________________________________________________ >>> _____ Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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-u >>> sers >>> http://www.freeswitch.org >> >> _____________________________________________________________________ >> ____ Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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-us >> ers >> http://www.freeswitch.org >> >> _____________________________________________________________________ >> ____ Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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-us >> ers >> http://www.freeswitch.org > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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-use > rs > http://www.freeswitch.org > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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-use > rs > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 !DSPAM:4f6c547e32761029518556!

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Re: [Freeswitch-users] Originate using inbound socket connection
by Peter Olsson 23 Mar '12

23 Mar '12

You need to put the other end of the call somewhere The correct string is (example) "originate sofia/gateway/test/1002 &park()" This will call 1002 and then park the call. /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Michael Lutz Skickat: den 21 mars 2012 14:49 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] Originate using inbound socket connection Hi Guys, I am working on a .net ESL host (inbound mode) and connect to the ESL using ESLConnection. Now I am trying to originate a call using the ESLConnection.Api() function. But it does't work as expected. It always returns: === Content-Type: api/response Content-Length: 125 Content-Length: 125 -USAGE: <call url> <exten>|&<application_name>(<app_args>) [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] === But I am pretty sure my string is right: {origination_caller_id_number=31341.....,origination_caller_id_name=31341......}sofia/external/31634258...(a)xxx.xxx.xxx.xxx I also tried with double {{ and }} but same result. Even if I take out the whole {} string and just use "sofia/external/etc.." I get the same message back. My code: ESLconnection eslDial = new ESLconnection("x.x.x.x", "8021", "x"); if (eslDial.Connected() == ESL_SUCCESS) { // Create a uuid used to identify the b-leg. string legb_uuid = eslDial.Api("create_uuid", "").GetBody(); string cDialString = "{{origination_uuid=" + legb_uuid + ",origination_caller_id_number=" + thisAni + ",origination_caller_id_name=" + thisAniName + "}}sofia/external/" + thisDestination; // Send the command var eslEvent = eslDial.Api("originate", cDialString); // Write the result to the console Console.WriteLine(eslEvent.Serialize(string.Empty)); return true; } Thanks for your help!, Regards, Michael Lutz _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 !DSPAM:4f69dabc32766734014540!

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VM sound very choppy
by Bzzz 23 Mar '12

23 Mar '12

FS last git fusionpbx =========== Hi list, I'm testing FS w/ twinkle, put iLBC as first CODEC and don't follow far end CODEC preference. All extensions have proxy_media set. When I call the VM (*97) I get a very choppy audio. FS log says: [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16@8000hz 1 channels 30ms when twinkle says: ilbc I guess that the source of my problem, but how can I change that for FS to switch to iLBC? (I already modified CODECs names & orders into conf/vars.xml to set iLBX@30i first, and twinkle is also set to 30ms) Jean-Yves -- Question authority.

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Call sent to group call terminates when group is busy instead of transferring
by Robert Longfield 22 Mar '12

22 Mar '12

I set up a group call for our support team in which all their phones ring when someone needs to speak with them. If they are busy the call should be transferred to a general extension which if not answered then goes to that extensions VM. My dialplan looks like: <extension name="group_dial_support"> <condition field="destination_number" expression="^2001$"> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="group/support@${domain_name}"/>  <action application="transfer" data="1000 XML default"/> <action application="hangup"/> </condition> </extension> What is happening is a caller selects the support option from the IVR, ever phone in the support group rings, which is what should happen. If no one picks up the call Freeswitch hangs up instead of transferring the call to extension 1000. You can see that I also tried to send the call directly to voicemail but that didn’t work either. The message I see when Freeswitch hangs up is: Channel sofia/internal/sip:1002@72.38.184.18:39042 hanging up, cause: USER_BUSY The full output from cli can be seen here: http://pastebin.freeswitch.org/18696 I would like to get the call to transfer properly. Thanks -Robert

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Remote reason header in xml cdr
by Paweł Pierścionek 22 Mar '12

22 Mar '12

Hi, I am originating new calls from local FS box via sofia to a remote SIP gw. Origination is directly via Socket or using LUA. Using fresh git version. Remote side responds ------------------------------------------------------------------------ recv 726 bytes from udp/[x]:15060 at 15:50:02.531445: ------------------------------------------------------------------------ SIP/2.0 503 Service Unavailable ... Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION" But local FS reports cause=41 into the CDR via mod_xml_cdr. Tried number of combinations of continue_on_fail, hangup_after_bridge, sip_ignore_remote_cause to see if it helps. My goal is to get the remote Q.850 cause into the local CDR. I cannot get/set session's variables if the session fails to connect so I cannot access sip_h_Reason and get it into the cdr :( Any ideas ? Urtho,

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DTMF passthrough delay / pass_rfc2833 problems
by Dmitry Sytchev 22 Mar '12

22 Mar '12

Hi all I'd like to know is there a way to make freeswitch pass RFC2833 dtmf right after it receives first packets as in pass_rfc2833 mode, but still recognize DTMF for bind_meta_app or bind_digit_action? Seems when i enable pass_rfc2833, bind_meta_app stops working. When we use ATA endpoints like Linksys PAP2T or SPA8000 without pass_rfc2833, ATAs sends little piece of inband DTMF followed by RFC2833 packets. While inband piece is immediately forwarded by FS, RFC2833 packets get relayed only after receiving end packets from endpoint, or at least delayed, effectively making double DTMF on other side. We can't use no-media or proxy-media mode as we need to deal with in-call features activated by DTMF :( What can be done to solve this issues? -- Best regards, Dmitry Sytchev, IT Engineer

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