[Serusers] Problems with PSTN and REINVITE

sip sip at arcdiv.com
Fri Apr 14 20:47:32 CEST 2006


I appear to be having problems with REINVITE messages lately (I may have
simply never before received them and there was ALWAYS a problem... I don't
know for certain). When I dial certain numbers -- so far it's only been
numbers over the PSTN going through likely Asterisk gateways, everything goes
fine UNLESS I get a reinvite. At that point, audio stops (though the stream
appears to remain open). 

My UA will send an INVITE, I'll hear the answer on the other side for about a
second -- audio will be fine... then a REINVITE is sent from the remote side
and audio stops completely while the stream remains active. 

Has anyone run across this before? Is there something special I should be
looking for in my ser.cfg to handle REINVITES as they bounce along? Is it a
problem, perhaps, with a swallowed ACK somewhere? 

I'm hoping someone can give me a clue or two to point me in the right direction. 

Thanks, 

N.




More information about the sr-users mailing list