[asterisk-users] Moving from res_sip to pjsip and simple bridge

[Joshua Colp]

On Thu, Feb 22, 2018, at 9:18 AM, Michele Pinassi wrote:
> Hi all,
> 
> on my old Asterisk 14.x box i use queue for some offices. For example,
> in this scenario phone 5710 is ringing (after passing through a
> queue...) and 5349 answer using REFER:
> 
>   -- SIP/5349-00000072 answered Local/SIP-5710 at MemberConnector-00000031;2
>     -- Local/SIP-5710 at MemberConnector-00000031;1 connected line has
> changed. Saving it until answer for SIP/5002-0000006e
>     -- Local/SIP-5710 at MemberConnector-00000031;1 answered SIP/5002-0000006e
>     -- Channel SIP/5349-00000072 joined 'simple_bridge' basic-bridge
> <a17ef15c-83a9-4fda-8a11-86ca653921e1>
>     -- Channel Local/SIP-5710 at MemberConnector-00000031;2 joined
> 'simple_bridge' basic-bridge <a17ef15c-83a9-4fda-8a11-86ca653921e1>
>     -- Stopped music on hold on SIP/5002-0000006e
>     -- Channel Local/SIP-5710 at MemberConnector-00000031;1 joined
> 'simple_bridge' basic-bridge <55d1ecc5-b251-4e8b-b0f3-ea08254c0ffd>
>     -- Channel SIP/5002-0000006e joined 'simple_bridge' basic-bridge
> <55d1ecc5-b251-4e8b-b0f3-ea08254c0ffd>
>        > 0xa081718 -- Probation passed - setting RTP source address to
> 172.20.xx.xx:60640
> 
> on new Asterisk 15.2 i decide to move to PJSIP but this functionality
> don't work and, on REFER, call dropped.
> 
> Maybe there's something needs to be enabled or checked ?

I don't understand the specific scenario here you are referring to with the REFER. A call is answered using a 200 OK sent back by the called party. Can you clarify further?

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org