My horror story has to do with SIP re-invites and was bugging me for at
least a couple of weeks.
I have my server connected by SIP to a few DID providers in different
places, and a Cisco router that serves as connection to PSTN in the 416
area. The Cisco router has special ACLs that allow connections only from my
Asterisk server. The problem was like this: when I made calls between PSTN
and my server, all was working OK When I made calls from a DID provider to
my server it was also working OK. But when I tried to route a call between a
DID provider to the Cisco router, there was no audio. That's because
Asterisk would re-invite the 2 peers to talk to each other directly but the
ACLs on the Cisco router would block the RTP packets.
The solution was obviously the "canreinvite=no" parameter for the all the
SIP peers.
