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