23 Feb
2009
23 Feb
'09
2:14 p.m.
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
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--
Daniel-Constantin Mierla
http://www.asipto.com