Re: [asterisk-users] Help in oh323 Gatekeeper
On Tue, Jul 14, 2009 at 02:10:47PM +0530, velusamy velu wrote: Dear All, I have installed GNU gatekeeper in my machine. I tested the calls using gatekeeper successfully. Now I have tried to Disable the gatekeeper in oh323.conf file gatekeeper=DISABLE Now I have tried to call, but the connection is not established. I have got following warning message in console. WARNING[8446]: chan_oh323.c:3555 cleanup_h323_connection: Call 'ip$192.168.8.96:30005/27890-f5194af7' not found (clear). What version of Asterisk is it? Any chance of using the built-in chan_h323? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help in oh323 Gatekeeper + does not know what to do when bridging the call
Actually I am facing a problem with H.323 (the standard and the ooh323) with Asterisk vesion 1.4.25 and I discover the following: 1) Using the standard h323 that come with Asterisk: The chan_h323.so it is not existed in the /usr/lib/asterisk/modules after doing the compilation and installation for (pwlib, openh323, /chanels/h323, asterisk), although make menuselect was done and the h323 channel was selected. Also, I was doing make opt for the h323, pwblib, and openh323. But the chan_h323.so is not existed. But when the h323 IP Phone originate the call, asterisk receive it (I see this in asterisk consol) and respond in the default context (even we added the [ ] in the h323.conf), but no voice at all (we do not hear the played wave file), I beleive this is because chan_h323.so is not existed under the /usr/lib/asterisk/modules/, maybe ! Why chan_h323.so is nto generated there? Why asterisk does not take in consideration that we added a friend in there h323.conf file and it should be authorized using its h323 id, and should be routed using the configured context, but all of this is not happenning, why? But ofcourse the telnet for the port 1720 is working, that means the h323 port is under listenning case. 2) Using the asterisk-addons-1.4.8: The chan_ooh323.so become existed in the /usr/lib/asterisk/modules and when we place a call from the IP Phone, it come to asterisk (I see this in asterisk consol) and the call manipulated in the default context (even if we added the endpoint configuration in the h323.conf file as a friend and determine its context), and we hear the wave file when it is played (so we hear a voice), but ofcourse it is not taking into consideration the endpoint configuration in the h323.conf file, so no authorization and actually can not do anything, because the configuration of the endpoint in the h323.conf file is not taken. Is it related to the Asterisk 1.4.25? Or we are missing something in the configuration? Regards Bilal Dear All, I have installed GNU gatekeeper in my machine. I tested the calls using gatekeeper successfully. Now I have tried to Disable the gatekeeper in oh323.conf file gatekeeper=DISABLE Now I have tried to call, but the connection is not established. I have got following warning message in console. WARNING[8446]: chan_oh323.c:3555 cleanup_h323_connection: Call 'ip$192.168.8.96:30005/27890-f5194af7' not found (clear). What version of Asterisk is it? Any chance of using the built-in chan_h323? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help in oh323 gatekeeper
Dear; I would like to ask: when Asterisk was registering on the gnugk, both (asterisk and gnugk) were on the same hardware machine and same IP address? Can they be on the same IP address? In case they were on the same IP address then: I am afraid the oh323 channel in asterisk will respond for the H323 endpoint (IP Phone) instead of the gnugk (specially if the IP Phone was in routed mode and not register to gnugk)? I mean, if the IP Phone was need to place call via the gnugk in the routed mode, and the call need to be send for Asterisk, so how can u avoid that oh323 channel in asterisk from responding for the IP Phone instead of the gnugk it self? Because if u let the IP Phone send the call for the IP address that asterisk running on it, then the h323 channel in the asterisk will respond as u know, so how to let the gnugk respond and not the asterisk h323 channel? Regards Bilal -- Dear All, I have installed GNU gatekeeper in my machine. I tested the calls using gatekeeper successfully. Now I have tried to Disable the gatekeeper in oh323.conf file gatekeeper=DISABLE Now I have tried to call, but the connection is not established. I have got following warning message in console. WARNING[8446]: chan_oh323.c:3555 cleanup_h323_connection: Call 'ip$192.168.8.96:30005/27890-f5194af7' not found (clear). Please any one give suggestions to disable the gatekeeper access in Asterisk... Thanks in Advance... Regards, Velusamy.K ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help in oh323 gatekeeper
On Tue, Jul 14, 2009 at 2:19 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear; I would like to ask: when Asterisk was registering on the gnugk, both (asterisk and gnugk) were on the same hardware machine and same IP address? Can they be on the same IP address? If I understand your questions: Can Asterisk and GnuGK both run as an h323 server on the same IP Address, the answer would be no unless they are running on different ports. You can not have two processes on the same machine/ip/port combination. In case they were on the same IP address then: I am afraid the oh323 channel in asterisk will respond for the H323 endpoint (IP Phone) instead of the gnugk (specially if the IP Phone was in routed mode and not register to gnugk)? I mean, if the IP Phone was need to place call via the gnugk in the routed mode, and the call need to be send for Asterisk, so how can u avoid that oh323 channel in asterisk from responding for the IP Phone instead of the gnugk it self? Because if u let the IP Phone send the call for the IP address that asterisk running on it, then the h323 channel in the asterisk will respond as u know, so how to let the gnugk respond and not the asterisk h323 channel? Right. If they both run on the same ip/port then the one started first would win, and listen for connections (the second app should fail to bind and complain). You could change the port, or the IP that the one of the apps is listening on. Hope that helps. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help with oh323
On Fri, 2006-02-10 at 00:45 +0500, Hussain Umair wrote: hi ive been tryin to get oh323 to work and installed it without any problems but it gives me the same error all the time this is the third time ive installed it..please if anyone can kindly help me out thanks in advance... [chan_oh323.so]Feb 10 00:35:29 WARNING[4891]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: _ZN11PTimedMutexC2Ev Feb 10 00:35:29 WARNING[4891]: loader.c:440 load_modules: Loading module chan_oh323.so failed! I had the same problem before. Have you checked the variable ASTERISKINCDIR in the Makefile from the oh323 path? It must be pointing to the path containig the include source of your running asterisk version. # # Set ASTERISKINCDIR variable to the directory containing the include files of # Asterisk PBX. # #ASTERISKINCDIR=/usr/src/asterisk/include ASTERISKINCDIR=/usr/src/asterisk-1.2.4/asterisk-1.2/include Regards, Umair. _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with Oh323
Theres a less than sign missing it should be: patch -p1 ../asterisk-oh323-0.7.2-pre1/openh323_1.13.5-make.patch (That will stop it hanging at least, not sure about anything else with H323) --Rob -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Federico Alves Sent: Sunday, June 12, 2005 8:47 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help with Oh323 I am blocked on the most simple step when compiling Oh323: Cd /openh323 patch -p1 ../asterisk-oh323-0.7.2-pre1/openh323_1.13.5-make.patch it hangs for ever and never finishes in my machine with Red Hat Enterprise Linux 3, fully updated. I need to use Asterisk HEAD and I therefore I want to use asterisk-oh323-0.7.2-pre1. Additionally, because it is the most debugged version. In the past, if I send an Oh323 call to an inexistent or dead IP address, the Dial functions hangs and the * does not allow the customer to disconnect, which made me lose a good client for my pre-paid app. Any help from the Oh323 gurus is appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP!!!! Ringback oh323
Specify option 'r' to dial application. Michael On Friday 01 August 2003 07:13 pm, [EMAIL PROTECTED] wrote: Hi What command i need to use to make a call with oh323 and hear the ringback sound Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
