[asterisk-users] No rtp activity

[Grygoriy Dobrovolskyy]

2009/3/1 michel freiha <michofr at gmail.com>

> Dear David,
> I'm using G729 pass though mode...No transcoding is used here
> Regarding concurrent calls, I have 3 asterisk servers working in load
> balancing mode...The issue that the same problem appear on 3 asterisk...each
> asterisk handle around 150 calls...
>
> I'll use tcpdump next time I face such issue
>
> Regards
>
>
>
> On Sat, Feb 28, 2009 at 7:21 PM, michel freiha <michofr at gmail.com> wrote:
>
>> Hi all....
>> I'm using asterisk for making PSTN calls from extensions registered on
>> OpenSIPS...In peak hours ,number of calls Increase dramatically to a non
>> logic number..When checking the calls using asterisk CLI I saw a lot of
>> calls in ringing status and after 300s(rtphold timeout), asterisk release
>> all calls...I checked the log file and found..
>> [Feb 28 11:34:14] NOTICE[19197] chan_sip.c: Disconnecting call
>> 'SIP/netcafe2-b7da99b8' for lack of RTP activity in 301 seconds
>> After that the log show:
>> [Feb 28 11:41:12] WARNING[19197] chan_sip.c: Remote host can't match
>> request CANCEL to call '6697777b27bb46ca01dc42b526adf7bd at Asterisk_IP_Address'.
>> Giving up.
>>
>> Did someone faced this issue before?
>>
>> Thanks for help
>>
>> Regards
>>
>
>
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Check your  opensips config, besides network check, you need to be sure that
your sip messages are not redirected to wrong servers, sometimes, if it the
case some call legs are messed up, and wrong cancels go to wrong servers...
I saw that with bad NAT config on opensips it was passible.
Good Day
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