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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090419/61e8a326/attachment-0001.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 19863 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090419/61e8a326/attachment-0001.jpeg
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 > > > > > > > ------------------------------ > This message was enriched by Impactia Technologies Ltd. > www.impactia.com > Please do not enrich<http://impactia.bezeqint.co.il/exclude.ASP?email=asterisk-users at lists.digium.com&domain=bezeqint.co.il&id=777>emails sent to me. > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090419/36b576f3/attachment.htm
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