Hi, I can see you've tried calling route[NATMANAGE] just before the route[TOVOICEMAIL] ! and that didn't work. Can you paste your configuration as well as a SIP trace for a voicemail call ! some logs of the same calls will help too.
Regards, Sammy On Wed, May 9, 2012 at 9:10 PM, <x-kamai...@sidell.org> wrote: > Greetings, > > Here's another problem I'm having with kamailio 3.2 and the standard > kamailio.cfg script. > > If the calling device is NATed, everything works fine if the original > call gets connected. That is, the INVITE sent to the called device has > the correct NAT fixups applied. > > But if the called device fails to answer and the script runs > route[TOVOICEMAIL], the call connects, but the INVITE sent to the > voicemail server doesn't have the NAT fixup applied. The result is > that the audio is connected in only one direction. > > It would appear that some rtpproxy function needs to get called to > apply the fixups prior to sending the INVITE to the voicemail server. > I've tried adding calls to route(NATMANAGE) at various places, but to > no avail. > > Any ideas? > > -- > Mark Sidell > Partner > Forte, Inc. > 919-942-7068 > fax 919-969-2844 > www.forteinc.com > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >
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