[asterisk-users] FW: issue with sip 180 responses
Nir Levi
nir.levi at gmail.com
Sun Apr 26 04:48:00 CDT 2009
Hello,
It's happens around 40 calls and above …
The **machine** accepts number of invites(we can see by tcpdump ) , but
asterisk sees part of them (we can see by CLI log) , and when it does ,
asterisk accepting an invite it reply the initator. (as it should ) – but
the rest of invites are just ignored.
it's seems like an O/S issue, because on asterisk level I can all going
correctly via logs (invite is accepted=> packet is generated=> and 180 is
sent immediately to the initiator …)
which tools can help me check kernel issues ?
Also,
tried to increase udp buffer (sysctl -w net.core.rmem_max=8388608) , but
seems the problem still persists.
Also , here a screenshot of a typical dump from network interface, you can
clearly see what's going on.
http://img7.imageshack.us/img7/6578/sip.png
Thank in advanced , Nir.
*C. Savinovich***
did you isolated the issue? , checked firewall , interface errors , routing
, sniffed the interface….
also , why using h323 and not IAX2 ?
*From:* asterisk-users-bounces at lists.digium.com [mailto:
asterisk-users-bounces at lists.digium.com] *On Behalf Of *C. Savinovich
*Sent:* Sunday, April 19, 2009 5:02 PM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* Re: [asterisk-users] issue with sip 180 responses
I am having a similar issue. Asterisk does not show ringback tone and I
investigated this due to it not reading sip invite 180. (or supposedly not
receiving it).. My solution is that now I am using h323
(ver 1.4.19)
CS
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090426/25b130b7/attachment.htm
More information about the asterisk-users
mailing list