Hi Premalatha, have you checked the opensips logs for error? can you post a call flow showing exactly which step fails ?
Regards, Bogdan Premalatha Kuppan wrote: > Hi, > > I posted before my query; but no repsonse :( > > Can some1 helps, whether this logic works, > > Iam using opensips 1.6.2(TLS) and Asterisk(1.4.3.1). > > 1. all users registering @ opensips > 2. All calls to opensips forwarded to sasterisk for IVR; thorugh IVR > input destination is known. > 3. Since asterisk 1.4x version doesnt support TLS/TCP. Iam forwarding > call to Opensips; indeed here some messy happens. Asterisk sents > invite to Opensips, opensips reach the destination via asterisk (First > of all, iam not sure this logic is true). For TLS, call is successful > but no audio. For TCP, call fails after IVR. > > Only my destination is TLS/TCP enabled. > > I appreicate valuable input and advice on thsi logic. > > Please help. > > Thanks, > Prem > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 20 - 24 September 2010, Frankfurt, Germany www.voice-system.ro _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
