[asterisk-users] h323 how to set it up?
Hi all, I have some questions about h323. Is it mandatory to install a oh323 or I can do h323 calls without patching or adding any new drivers ti asterisk? I did compile the asterisk with channel driver chan_h323 but it still gives me this error: [Feb 28 18:12:58] WARNING[1902]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'H323' (cause 66 - Channel not implemented) what shoul I do to have it implemented? Can somebody recommend some references on how to set up h323 ? Thx, Igor This message was scanned by Barracuda Networks ___ --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] SIP??
ip_pbx2 is not asterisk, it knowk only PCMU,PCMA,g723,g729 On Thursday 08 February 2007 19:00, Vicky wrote: config problem . what pbx does ip_pb2 runs ? ( is it asterisk ? ) in peer definition try allowing all codecs .. ( gsm , ulaw,alaw,ilbc ) On 08/02/07, Florea Igor [EMAIL PROTECTED] wrote: Hi, I'm new to *,so i apologize for stupid questions. I'm having problem with this arhitecture: I'm calling asterisk from behind a NAT(sjphone user) with a low band so I'm using GSM codec. In extensions.conf I have: exten = 337,1,Dial(SIP/99@ip_pbx2) so when i dial 337 from sjphone Asterisk is colling 99 on ip_pbx2. RTP stream between sjphone and Asterisk are ok (GSM). The problem is rtp packets from Asterisk to ip_pbx2 are also GSM although ip_pbx2 sip is telling asterisk It only knows codec 0 Is this a config problem or a bug? Igor, ___ --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 -- There are 10 kinds of people in the world: those who know binary and those who don't. Igor Florea Ing. dezvoltare Phone: +40 21 232 04 24 Fax: +40 21 232 31 56 Local time: GMT+2 www.topex.ro ___ --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
[asterisk-users] SIP??
Hi, I'm new to *,so i apologize for stupid questions. I'm having problem with this arhitecture: I'm calling asterisk from behind a NAT(sjphone user) with a low band so I'm using GSM codec. In extensions.conf I have: exten = 337,1,Dial(SIP/99@ip_pbx2) so when i dial 337 from sjphone Asterisk is colling 99 on ip_pbx2. RTP stream between sjphone and Asterisk are ok (GSM). The problem is rtp packets from Asterisk to ip_pbx2 are also GSM although ip_pbx2 sip is telling asterisk It only knows codec 0 Is this a config problem or a bug? Igor, ___ --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
