[asterisk-dev] No rtp activity

[michel freiha]

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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