hello thanks Dmitry for your useful hints. i enable debug and solve my problem:). it was codec compatibility problem. but it is so strange; if i set codec g711alaw in cisco router and asterisk, i have the mentioned problem but if i set codec to transparent in cisco router, every thing will be ok. is there any difference between g711 codecs which cisco and asterisk utilize?
On Wed, May 6, 2015 at 11:56 AM, Dmitry Melekhov <[email protected]> wrote: > 06.05.2015 10:58, s m пишет: > > Hello! > > I'm not h323 expert, may be somebody else can understand from this log > what is happening, but I can't :-( > > Could you, please, provide log with > > tracelevel=6 > > in ooh323.conf ? > > Thank you! > > hello Dmitry >> >> thank you for your reply. Ok, you are right. i want to configure trunk >> h323 between asterisk 11.13.1 and 2800 cisco router. this is my scenario: >> >> PBX(100)--->cisco--->asterisk11.13.1---->PBX(200) >> >> when i call from 100 to 200, everything is ok but when i call from 200 to >> 100, phone rings but after i answer it, i have no voice and call terminates >> after 5 seconds. this is ooh323 debug(in asterisk11.13.1 system): >> >> ooh323_get_rtp_peer OOH323/peer-2-5 -> (null):0, 1 >> >> >> > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
