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:
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK, UPDATE
This 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.
Important note: PJSIP is NOT a FreePBX addition - it is part of the underlying Asterisk implementation. There are FreePBX management components, but the actual underlying PJSIP stuff is provided by someone else.
My provider has told me that they do not support PJSIP as an interface. There are too many problems that Chan-SIP simply doesn’t have. The litany of problems with PJSIP is simply too overwhelming to make it the interface model of choice right now.
Several people have noted that PJSIP doesn’t appear to be ready-for-primetime. In these fora, I have voiced the opinion that, like IAX, PJSIP is really not intended to be a broadly used tool. For connections to VOIP providers, Chan-SIP is still the way to go. For connections to individual phones, PJSIP might be the right tool, but for many phones (specifically older SIP devices) Chan-SIP is still the interface of choice.
In another couple years, PJSIP may well have progressed to the point that it is as universally applicable as Chan-SIP. For right now, though, I recommend against it for anything but individual phones.
I realize PJSIP is Digiums’s domain, hence my reference to the documented issue. In the scope of Asterisk, PJISP is the mainstream channel driver and chan-SIP is considered legacy. That said I agree there’s work to be done in the PJSIP arena and currently I am only utilizing in test environments with a few different providers. PJSIP has come along way since implemented in Asterisk 12 and does get a bad rap due to the learning curve associated with the configuration. Obviously the FPBX team isn’t responsible for a fix, but FPBX does support the implementation of PJSIP and I am simply conveying the issue to this forum.