> 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
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