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

[Michele Pinassi]

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 ?

Michele


-- 
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - centralino at unisi.it

Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo, http://www.faq.unisi.it 


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