[asterisk-users] Call drop weirdness
Brett Lehrer
Brett.Lehrer at solarismed.com
Tue Oct 23 12:02:14 CDT 2012
> I'm running Asterisk 10.7.0 with three sip trunks to my call termination provider. For the most part everything works great.
>However, at apparently random times and usually about 20 mins or so into the call, the outbound audio stream dies.
>The call stays connected and the inbound audio works fine. The thing is, it happens on such an irregular basis
>(once or twice per day) that I can't get a data dump to see what actually happens. Some times there is a bit of artifacting
>which takes place just prior to the drop, but mostly
>nothing: it just drops.
>I've checked and rechecked firewall settings. Bandwidth consumption on the Inet link varies, but the dropped audio
>happens even on off-peak times.
>I'm considering giving the Asterisk box a public IP on one IF and bypassing the FW to rule out NAT weirdness.
>Any thoughts on things to look at would be greatly appreciated.
>Kind Regards,
>Chris
I'm not sure if this is any help, but I had a familiar issue to this, except it involved transferring to another internal extension.
The symptoms were the same though. Only outbound audio would cut out and it was very sporadic (~10% of transfers).
The issue ended up being with the trunking service and their spotty support with UPDATE messages. We had to disable
rpid_update in sip.conf and a couple other bits that I can't offhand remember. I'd check with the trunk provider on the issue.
Best of luck,
Brett Lehrer
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121023/fb190ccf/attachment.htm>
More information about the asterisk-users
mailing list