[asterisk-users] issue with sip 180 responses

[C. Savinovich]

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

 

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Yehavi
Bourvine
Sent: Sunday, April 19, 2009 4:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] issue with sip 180 responses

 

Maybe it is buffering issues with the kernel? Does it happen only when there
is a peak in the new calls rate? Do all 180 messages get dropped?

 

                           __Yehavi:

 

2009/4/19 Nir Levi NirL at bezeqint.co.il



 


 

Hello,

SIP invites are accepted from imitator , but 'SIP 180'  is not  responded
back  to imitator. 

 

By inspecting the issue , we can *see* the response is generated and sent
from asterisk (via asterisk logger ("sip debug" )) , but while sniffing the
interface with tcpdump, we can't see 180 response on the interface.

 

We don't have errors on the interface, firewall is disabled , seems there's
no packet lost  (checked with ping with low interval ) , and routes are ok .


 

By our tests we can see there is a direct connection between the mass of the
calls, and between the lost of sip 180 responses.

 

We're using Asterisk 1.4.4, with real-time configuration, also we made few
*modifications* in asterisk source code (changed app_dial.c, /app_macro.c,
/func_cdr.c).

 

We're not sure if the problem on the OS Level (Centos 5.2) or in the
asterisk application.

Please assist.

~Nir

  

 

 


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