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 <a17ef15c-83a9-4fda-8a11-86ca653921e1> -- Channel Local/SIP-5710@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@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 - central...@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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