Hello,
I have a carrier who provides PSTN gateway services. They have
multiple redundant sip gateway devices in their network. The problem
occurs when one of their devices starts to have issues. I will
receive an INVITE request from both gateways with the same call-id.
The problem is that my Kamailio system doesn't detect that I already
set a call up for the INVITE once, and forwards the request to another
server in the dispatcher list. What I end up with is a call on two
asterisk servers, but only one has the actual RTP stream. The BYE
request gets routed to the wrong server, and everything just gets
screwy. If anyone could provide any hint on how I might be able to deal with
this scenario, I would really appreciate it.
I have attached my current config file, and the following is a link to
a google spreadsheet which
shows the SIP trace.
http://spreadsheets.google.com/ccc?key=pU5i2J6Ck3b519-_M6Et3cw
I have masked my IP addresses for my own sanity.
XX.XX.XX.179 - Kamailio SIP Gateway
XX.XX.XX.189 - Asterisk1
XX.XX.XX.186 - Asterisk2
Thanks!
Geoff