Sorry about the cross post. I wasn't sure how many people were on both the OpenSIPs and Kamailio mailing lists... and since this is a 'core' issue, I figured it would be good to get input from the most people. In the future I will only post to one.
I will go down the path of the htable, what kind of performance/memory hit am I going to take? This system has a lot of memory available to it, how would I increase memory appropriately to ensure the htable had enough to live happily? thanks, Geoff On Mon, Feb 23, 2009 at 7:14 AM, Daniel-Constantin Mierla <mico...@gmail.com> wrote: > Hello, > > please do not cross-post on many mailing lists. Will create confusion about > available solutions. > > Theoretically, this is valid in SIP (e.g., 2 invites with same call-id) -- > it is same scenario as parallel forking in upstream. > > However, if you know that this shouldn't happen, you can try to fix it from > config. > > Fist is to identify why the BYE is routed to the wrong server. It should > follow the Route set and contact address. Can you provide the pcap file of > such call? > > As solution to deny new invites with same call id is to use the htable > module. Set a key there based on call id (eventually plus from user, from > tag, etc.) and check it before processing the invite, if there is one, drop > it. > > You just set key auto-expire for 30-60sec so it gets automatically deleted. > > Note that htable is in devel version (upcoming 1.5.0), but should work out > of the box with 1.4: > http://kamailio.org/docs/modules/1.5.x/htable.html > > Cheers, > Daniel > > > On 02/22/2009 03:24 AM, Geoffrey Mina wrote: >> >> 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 >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users@lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > _______________________________________________ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
