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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoPlainText>> I'm running Asterisk 10.7.0 with three sip trunks to my call termination provider. For the most part everything works great.<o:p></o:p></p><p class=MsoPlainText>>However, at apparently random times and usually about 20 mins or so into the call, the outbound audio stream dies. <br>>The call stays connected and the inbound audio works fine. The thing is, it happens on such an irregular basis <o:p></o:p></p><p class=MsoPlainText>>(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<o:p></o:p></p><p class=MsoPlainText>>which takes place just prior to the drop, but mostly<o:p></o:p></p><p class=MsoPlainText>>nothing: it just drops.<o:p></o:p></p><p class=MsoPlainText><o:p> </o:p></p><p class=MsoPlainText>>I've checked and rechecked firewall settings. Bandwidth consumption on the Inet link varies, but the dropped audio <o:p></o:p></p><p class=MsoPlainText>>happens even on off-peak times.<o:p></o:p></p><p class=MsoPlainText><o:p> </o:p></p><p class=MsoPlainText>>I'm considering giving the Asterisk box a public IP on one IF and bypassing the FW to rule out NAT weirdness.<o:p></o:p></p><p class=MsoPlainText><o:p> </o:p></p><p class=MsoPlainText>>Any thoughts on things to look at would be greatly appreciated.<o:p></o:p></p><p class=MsoPlainText><o:p> </o:p></p><p class=MsoPlainText>>Kind Regards,<o:p></o:p></p><p class=MsoPlainText>>Chris<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>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. <o:p></o:p></p><p class=MsoNormal>The symptoms were the same though. Only outbound audio would cut out and it was very sporadic (~10% of transfers).<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>The issue ended up being with the trunking service and their spotty support with UPDATE messages. We had to disable <o:p></o:p></p><p class=MsoNormal>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.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Best of luck,<o:p></o:p></p><p class=MsoNormal>Brett Lehrer<o:p></o:p></p></div></body></html>