Re: [asterisk-users] Conference problem
The CM is sending the BYE messages. Any ideas? Christian --- On Wed, 4/22/09, Martin asteriskl...@callthem.info wrote: From: Martin asteriskl...@callthem.info Subject: Re: [asterisk-users] Conference problem To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, April 22, 2009, 8:08 PM run a sip debug and check whether it's asterisk disconnecting the calls (usually a SIP BYE message) or whether Asterisk is getting the disconnect from your Cisco GW Martin On Wed, Apr 22, 2009 at 10:56 AM, Cristi Iconaru cristi_icon...@yahoo.com wrote: Hello all, I have some issues with the MeetMe application. The working topology is as follows. The Asterisk (1.4.22-rc5) is connected through SIP trunk to a Call Manager (6.1.2) which is connected to a Cisco Voice Gateway. The Gateway is connected to PSTN through a PRI. The calls are forwarded to Asterisk by the CM. The problem is that some users who are calling in from PSTN are getting disconnected from the conference room after a period of time. They can get in but after a while suddenly they are disconnected. The funny thing is that on the Asterisk CLI/logs no errors/retrans/etc. appeared. The Asterisk has no Zaptel hardware. All the necesary modules are installed. Thanks, Christian ___ -- 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 ___ -- 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 ___ -- 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
[asterisk-users] Conference problem
Hello all, I have some issues with the MeetMe application. The working topology is as follows. The Asterisk (1.4.22-rc5) is connected through SIP trunk to a Call Manager (6.1.2) which is connected to a Cisco Voice Gateway. The Gateway is connected to PSTN through a PRI. The calls are forwarded to Asterisk by the CM. The problem is that some users who are calling in from PSTN are getting disconnected from the conference room after a period of time. They can get in but after a while suddenly they are disconnected. The funny thing is that on the Asterisk CLI/logs no errors/retrans/etc. appeared. The Asterisk has no Zaptel hardware. All the necesary modules are installed. Thanks, Christian ___ -- 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] Conference problem
run a sip debug and check whether it's asterisk disconnecting the calls (usually a SIP BYE message) or whether Asterisk is getting the disconnect from your Cisco GW Martin On Wed, Apr 22, 2009 at 10:56 AM, Cristi Iconaru cristi_icon...@yahoo.com wrote: Hello all, I have some issues with the MeetMe application. The working topology is as follows. The Asterisk (1.4.22-rc5) is connected through SIP trunk to a Call Manager (6.1.2) which is connected to a Cisco Voice Gateway. The Gateway is connected to PSTN through a PRI. The calls are forwarded to Asterisk by the CM. The problem is that some users who are calling in from PSTN are getting disconnected from the conference room after a period of time. They can get in but after a while suddenly they are disconnected. The funny thing is that on the Asterisk CLI/logs no errors/retrans/etc. appeared. The Asterisk has no Zaptel hardware. All the necesary modules are installed. Thanks, Christian ___ -- 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 ___ -- 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] conference problem without zapata interface
Hello, On Thu, 2003-07-17 at 09:00, Andrzej Radke wrote: In file app_meetme.c we can read A ZAPTEL INTERFACE MUST BE\n INSTALLED FOR CONFERENCING FUNCTIONALITY.\n I receive message, when I try conference WARNING[28686]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' Does it means that I cannot establish conference without any hardware zaptel interface ??? No. What can I do if I want make conference only between my sip phones using asterisk ?? Buy it ??? Yes. Alternatively, you get the zaptel drivers, edit the Makefile to build 'ztdummy' (remove the '#' before ztdummy on the line just after the line starting with MODULES), compile, install, and do modprobe ztdummy. Why this would help can be found in the archives (just as this answer) and is left as excercise for the reader. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] conference problem without zapata interface
You need to load the ztdummy kernel driver. It will provide the pseudo timing needed to sync the conference channel. It is a driver that creates a dummy Zaptel hardware interface. quote who=Andrzej Radke Hello ! In file app_meetme.c we can read A ZAPTEL INTERFACE MUST BE\n INSTALLED FOR CONFERENCING FUNCTIONALITY.\n I receive message, when I try conference WARNING[28686]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' Does it means that I cannot establish conference without any hardware zaptel interface ??? What can I do if I want make conference only between my sip phones using asterisk ?? Buy it ??? Greeting Andrzej Radke ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
