Hi all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, I'll make a call, ore receive a call, and there will be NO sound. The strange part is that both endpoints are public IP addresses so NAT isn't in play and a sniffer trace reveals that the packets simply aren't being sent. It only seems to happen on a particular trunk. The same phone calling on a different trunk works just fine. Any ideas? -- Take care and have fun, Mike Diehl.
2010/10/26 Mike Diehl <mdiehl at diehlnet.com>> Hi all, > > I seem to be having a strange problem with a sip trunk. > > On a fairly frequent basis, I'll make a call, ore receive a call, and there > will be NO sound. The strange part is that both endpoints are public IP > addresses so NAT isn't in play and a sniffer trace reveals that the packets > simply aren't being sent. > > It only seems to happen on a particular trunk. The same phone calling on a > different trunk works just fine. > > Any ideas? >codec incompatibilities ? t.38 ?> -- > > Take care and have fun, > Mike Diehl. > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101026/7d5ab4ae/attachment.htm
Do you have canreinvite=yes anywhere? If yes, try setting it to no. Also pasting your sip.conf here would be helpful. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-27 6:16 PM, "Mike Diehl" <mdiehl at diehlnet.com> wrote: There are NO ACL's in place, either at the network level, or application level. We have a public address, so as far as I know, there are no forwarding rules in place. On Wednesday 27 October 2010 4:04:16 pm Philipp von Klitzing wrote:> Hi! > > > I've turned off t....Take care and have fun, Mike Diehl. -- ___________________________________________________________... -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101027/b0d11d63/attachment.htm