On 04/07/11 03:00, Shariq Khan wrote:
I am facing one way audio problem in sip trunking between asterisk and avaya.+-------------+ +----+ | avaya sip |-------| P1 | +-------------+ +----+ | | | +-------------+ | Asterisk | WAN ------------------------------------------------- | | LAN +-------------+ | / +----+ / | P2 |--+ +----+ When P1 dial P2, P2 hears voice clear but P1 could not hear any voice. My sip.conf is [avaya] type=peer fromdomain=xx.xx.xx.xx host=xx.xx.xx.xx disallow=all allow=ulaw dtmfmode=rfc2833 canreinvite=yes -- Regards, Shariq Khan 0333-3501125 -- _____________________________________________________________________ -- 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
Turn off reinvite on all extensions and SIP trunks involved and try again. Lyle Giese LCR Computer Services, Inc. -- _____________________________________________________________________ -- 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
