@shauberg wrote:
When configuring a PJSIP trunk with a provider that uses SIP INFO requests for keepalive purposes the call gets torn down by the provider after a number of SIP INFO requests which are not acknowledge.
In my case the INFO method does not appear in the SDP headers so possibly a provider issue:Invite request:
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFERInvite response:
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK, UPDATEThis is a documented issue with Asterisk 13: ASTERISK-24986, but I am wondering if anyone else is having this issue and if a possible work around has been sorted. Currently this is a deal breaker for PJSIP in this case and have resorted to configuring the offending trunks with chan_SIP.
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