[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
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