SIP Jitter Buffer and Sangoma A104 Question

I have a customer who has a Sangoma A104 (no echo canceler) on an asterisk server. I enabled sip jitter buffer and set it to adaptive to try to solve some jitter problems on some remote phones. When the jitter buffer is set to adaptive they have a problem with attended transfers on calls that come in through the Sangoma.

The call comes in and is answered. When the call is transfered to the second internal extension the person who answers can hear, but the external caller cannot. The last thing the external caller hears is MOH during the transfer.

I set jbimpl=fixed, and the problems seems to have gone away.

I’m not familiar with the Sangoma card (or zap trunks at all since I do all sip trunking), but I assume this card is using an old version of it’s firmware.

Has anyone seen this problem? Any suggestions?


A little more info on this. Today I noticed that when a transfer completes we start seeing a flood of these messages…

I’m finding references to this in older versions of asterisk. But, we’re runnign