- FreeSWITCH-users - FreeSWITCH Mailing Lists

2-party video call is not working
by Vincent Xia 19 Jun '13

19 Jun '13

hi all, im testing video features of FS including video record/playback, 2-party video call and video conference, everything works fine except the 2-party video call, after the call get connected i clicked "start" button of my eyebeam softphone to send video, the FS log displays 2013-06-18 17:54:16.346026 [DEBUG] switch_rtp.c:4686 Correct ip/port confirmed. 2013-06-18 17:54:16.346026 [DEBUG] switch_ivr_bridge.c:72 Send signal sofia/internal/sip:1006@192.168.0.100:58734 [BREAK] 2013-06-18 17:54:16.346026 [DEBUG] switch_ivr_bridge.c:73 sofia/internal/ 1002(a)192.168.0.100 video thread ended. while in video conference (which works fine), it's just 2013-06-18 17:54:16.346026 [DEBUG] switch_rtp.c:4686 Correct ip/port confirmed. i've had the following lines in vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM,H261,H263,H263-1998"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM,H261,H263,H263-1998"/> is there something wrong with my config?

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Re: [Freeswitch-users] Sangoma D500 mod_sangoma_codec
by Andre Demattia 18 Jun '13

18 Jun '13

Ds3 or T1. Anything to terminate sip traffic. Pretty much something that I can be my own sip provider. I'm not sure what equipment to get. I need 1000's if ports and 1000 cps. -----Original Message----- From: "Guillermo Ruiz Camauer" <grcamauer(a)gmail.com> Sent: ‎6/‎17/‎2013 9:56 PM To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Sangoma D500 mod_sangoma_codec By dial tone do you mean an analog line? Guillermo Sent from my iPhone On 17/06/2013, at 22:51, Ken Rice <krice(a)freeswitch.org> wrote: You mean something to gate SIP to the traditional TDM PSTN? (like T1s or E1s?) if so then you are looking for something like an A104DE (4 port T1 card) or something like an Adtran 900 series gateway if you want a hardware device On 6/17/13 8:40 PM, "Andre Demattia" <andretodd(a)verizon.net> wrote: Hmm I was looking for something to turn sip traffic to dial tone. I guess this isn't what I need. From: Guillermo Ruiz Camauer <mailto:grcamauer@gmail.com> Sent: ?6/?17/?2013 9:07 PM To: FreeSWITCH Users Help <mailto:freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] Sangoma D500 mod_sangoma_codec The simple andwer is NO. The Sangoma D500 is a transcoding card. It converts an RTP stream in g729 to PCMU for example. It does absolutely nothing with SIP signaling. It is recommended for high traffic systems or for a large number of G729 licenses. Note that the D500 can be shared among several servers. There are smaller cards like the D100 which is the one I have which run up to 120 simultaneous transcoding streams. A similar card is the Digium TCE400B. Guillermo Sent from my iPhone On 17/06/2013, at 20:07, Andre <andretodd(a)verizon.net> wrote: HI, has anyone used Sangoma D500 with freeswitch and if so how does it work? Does this card connected to a Gig E become a SIP Trunk? _________________________________________________________________________ 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://lists.freeswitch.org/mailman/options/freeswitch-users> 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch _________________________________________________________________________ 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@list [The entire original message is not included.]

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Call flow not getting back if transfer application execute
by bhavik patel 18 Jun '13

18 Jun '13

Hi All, I have created one Dialplan and tried to debug call flow and found some issues in that. 1) If we use transfer application for call handling then we are not getting call pointer back for further execution and execution stops at that point. 2) We're not getting call pointer back from transfer application in any case,Execution stops. If transfer application failed then it execute another bridige application will execute but transfer application can not back to next pointer : Here is the my Dialplan: <application app_name="set" app_data="hangup_after_bridge=true"></application> <application last_executed="true" app_name="transfer" app_data="4545664 XML default"></application> <application app_name="bridge" app_data="sofia/pbxsystem/test"></application> -- Thanks, Bhavik Patel

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Load all configurations from DB
by Sayyed Mohammad Emami Razavi 18 Jun '13

18 Jun '13

Is there any way to load all configurations from DB (MySQL) not from XML or a way to make xml response for FS reloading XML according to configurations?!

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Re: [Freeswitch-users] Sangoma D500 mod_sangoma_codec
by Andre Demattia 18 Jun '13

18 Jun '13

Yes thanks -----Original Message----- From: "Ken Rice" <krice(a)freeswitch.org> Sent: ‎6/‎17/‎2013 9:48 PM To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Sangoma D500 mod_sangoma_codec You mean something to gate SIP to the traditional TDM PSTN? (like T1s or E1s?) if so then you are looking for something like an A104DE (4 port T1 card) or something like an Adtran 900 series gateway if you want a hardware device On 6/17/13 8:40 PM, "Andre Demattia" <andretodd(a)verizon.net> wrote: > Hmm I was looking for something to turn sip traffic to dial tone. I guess this > isn't what I need. > > From: Guillermo Ruiz Camauer <mailto:grcamauer@gmail.com> > Sent: ?6/?17/?2013 9:07 PM > To: FreeSWITCH Users Help <mailto:freeswitch-users@lists.freeswitch.org> > Subject: Re: [Freeswitch-users] Sangoma D500 mod_sangoma_codec > > The simple andwer is NO. The Sangoma D500 is a transcoding card. It converts > an RTP stream in g729 to PCMU for example. It does absolutely nothing with > SIP signaling. It is recommended for high traffic systems or for a large > number of G729 licenses. Note that the D500 can be shared among several > servers. > There are smaller cards like the D100 which is the one I have which run up to > 120 simultaneous transcoding streams. > A similar card is the Digium TCE400B. > > Guillermo > > Sent from my iPhone > > On 17/06/2013, at 20:07, Andre <andretodd(a)verizon.net> wrote: > >> HI, has anyone used Sangoma D500 with freeswitch and if so how does it work? >> >> Does this card connected to a Gig E become a SIP Trunk? >> _________________________________________________________________________ >> 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://lists.freeswitch.org/mailman/options/freeswitch-users> >> 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Re: [Freeswitch-users] Sangoma D500 mod_sangoma_codec
by Andre Demattia 18 Jun '13

18 Jun '13

Hmm I was looking for something to turn sip traffic to dial tone. I guess this isn't what I need. -----Original Message----- From: "Guillermo Ruiz Camauer" <grcamauer(a)gmail.com> Sent: ‎6/‎17/‎2013 9:07 PM To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Sangoma D500 mod_sangoma_codec The simple andwer is NO. The Sangoma D500 is a transcoding card. It converts an RTP stream in g729 to PCMU for example. It does absolutely nothing with SIP signaling. It is recommended for high traffic systems or for a large number of G729 licenses. Note that the D500 can be shared among several servers. There are smaller cards like the D100 which is the one I have which run up to 120 simultaneous transcoding streams. A similar card is the Digium TCE400B. Guillermo Sent from my iPhone On 17/06/2013, at 20:07, Andre <andretodd(a)verizon.net> wrote: HI, has anyone used Sangoma D500 with freeswitch and if so how does it work? Does this card connected to a Gig E become a SIP Trunk? _________________________________________________________________________ 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

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Sangoma D500 mod_sangoma_codec
by Andre 18 Jun '13

18 Jun '13

HI, has anyone used Sangoma D500 with freeswitch and if so how does it work? Does this card connected to a Gig E become a SIP Trunk?

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Re: [Freeswitch-users] Build failure with libpq
by kellon.langdon＠sis-grenada.com 18 Jun '13

18 Jun '13

I managed to build using the Postgresql Repo, But then it broke freetdm couldn't get modules to load anymore so had to revert to compile without pgsql. If you're running on CentOS 6 then try yum update before compiling. From: Claus Andersen <clan(a)wheel.dk> Date: 06/17/13 14:24Subject: Re: [Freeswitch-users] Build failure with libpq On Fri, 14 Jun 2013, Claus Andersen wrote: > I am trying to make/build with core-pgsql but it fails: I have created FS-5520 to track this issue. Can anyone confirm that they have successfully linked against libpq in any fairly recent version? If you have linked to a fairly recent version did you use your disto's repo or the PostgreSQL repo? >From my understanding there has been no incompatible protocol changes to libpq since version 7.4 where version 3 of the protocol was introduced. >From what I can gather we do everything correct for the build: http://www.postgresql.org/docs/9.2/interactive/libpq-build.html Furthermore it does not look like the build error is directly related to libpq but I cannot suss out what actually is b0rken. Can anyone shed any light on that? Kind Regards, Claus Andersen _________________________________________________________________________ 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

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FreeSWITCH Weekly News and Notes
by Michael Collins 17 Jun '13

17 Jun '13

Greetings! We are all looking forward to our upcoming conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_06_19>for June 19th, because we are releasing the official ClueCon 2013 <http://www.cluecon.com> speaker schedule! Additionally, the FreeSWITCH core developer team will be on the conference call to hang out with us. Let's all be there this Wednesday and pack out the conference call! In addition to FreeSWITCH, the core development team also built the Cudatel Communication Server <http://www.cudatel.com>. On last week's conference call <http://wiki.freeswitch.org/wiki/FS_weekly_2013_06_12> we did a thorough tour of the CudaTel interface and its capabilities. We also talked about how the CudaTel can utilize external call control by utilizing an HTTAPI server. The CudaTel really is a great product and we highly recommend it for those enterprises that need a simple yet powerful IP communications platform. Thanks for your time and we'll talk to everyone on Wednesday! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Inconsistent Event CHANNEL_PROGRESS
by Ashish gautam 17 Jun '13

17 Jun '13

Hi, I have subscribed to CHANNEL_PROGRESS event which inconsistent i.e. sometimes it doesn't get generated. I am generating outbound call through event socket. -- Ashish Gautam Nucleus Microsystems (Pvt.) Ltd. Ph. 011 47574758

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