Moving From Res_sip To Pjsip And Simple Bridge

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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@MemberConnector-00000031;2
    — Local/SIP-5710@MemberConnector-00000031;1 connected line has changed. Saving it until answer for SIP/5002-0000006e
    — Local/SIP-5710@MemberConnector-00000031;1 answered SIP/5002-0000006e
    — Channel SIP/5349-00000072 joined ‘simple_bridge’ basic-bridge

    — Channel Local/SIP-5710@MemberConnector-00000031;2 joined
‘simple_bridge’ basic-bridge
    — Stopped music on hold on SIP/5002-0000006e
    — Channel Local/SIP-5710@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 ?

Michele

One thought on - Moving From Res_sip To Pjsip And Simple Bridge

  • 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?

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    Joshua Colp Digium, Inc. | Senior Software Developer
    445 Jan Davis Drive NW – Huntsville, AL 35806 – US
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