for which user/number sip reinvite is for? ooh! you are running a direct application without any dialplan or user, may be that is the cause! I think you should first write fax dialplan, reload asterisk and test again with originate but this time with extension not application.
Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ On Sat, Oct 8, 2011 at 12:20 AM, James Sharp <ja...@fivecats.org> wrote: > On 10/07/2011 12:27 AM, Nasir Iqbal wrote: > >> Check firewall and NAT settings! >> >> Monitoring sip and media flow from asterisk cli can help, use "sip set >> debug on", "rtp set debug on" and "udptl set debug on" >> >> > No NAT involved and I shut off IPTables. Still no luck. Debug shows the > SIP invite, RTP frames going in & out, the SIP reinvite, and then UDPTL > frames coming in until timeout. > > See the entire transaction at http://pastebin.ca/2087758 > > > -- > ______________________________**______________________________**_________ > -- 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<http://lists.digium.com/mailman/listinfo/asterisk-users> >
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