On Sat, Mar 13, 2010 at 11:28 AM, bhrugu mehta <[email protected]>wrote:
> Hi, all > i have used opensips as registrar. > my scenario, > > opensips->asterisk(routing logic)->opensips > > i have done with opensips to asterisk call . > asterisk deside where to call go , and if local call then go to opensips. > > asterisk to opensips call not done. > > any suggetion? > > post some network trace to suggest Ram, > -- > Bhrugu Mehta > Sr. S/W Engineer (D&D) > VOIP,Telephony Team > India > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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