What is happening:
call comes in to the tenant and they answer it
they transfer the call to another phone
audio goes dead both directions
We receive, " Got SIP response 500 "Internal Server Error" back from PROVIDERSIP:5060"
Provider indicates that they are receiving both a "reinvite" as well as an "update" to which they are responding 200 OK to both.
They said normally they receive one or the other but not both.
They said that "update" isn't technically supported yet so if we could turn that off, it should fix the issue.
I guess I have two questions for the forums:
Does anyone know how to turn off the "update" so I can test this theory?
Does anyone have any other ideas on this?
There are no firewalls or anything blocking anyone.
This is happening to two completely separate companies, the only thing they have in common is the same internet provider.
Any help would be greatly appreciated.
Polycom phones with one
Polycom phones with one company and Grandstream with the other company.
Asterisk is running 1.8
Last night we turned off qualify on the sip trunk and the system stopped sending the update at that point in the transfer and it seems to have resolved the issue.
I don't understand why qualify would do that at that moment, but it appears resolved at the moment.
I'll update this if the issue returns.
What type of Phones are you using? What version of Asterisk is this?
Whats the config for the trunk look like?