Re: [asterisk-users] Moving from res_sip to pjsip and simple bridge
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/5349-0072 answered Local/SIP-5710@MemberConnector-0031;2 > -- Local/SIP-5710@MemberConnector-0031;1 connected line has > changed. Saving it until answer for SIP/5002-006e > -- Local/SIP-5710@MemberConnector-0031;1 answered SIP/5002-006e > -- Channel SIP/5349-0072 joined 'simple_bridge' basic-bridge > > -- Channel Local/SIP-5710@MemberConnector-0031;2 joined > 'simple_bridge' basic-bridge > -- Stopped music on hold on SIP/5002-006e > -- Channel Local/SIP-5710@MemberConnector-0031;1 joined > 'simple_bridge' basic-bridge <55d1ecc5-b251-4e8b-b0f3-ea08254c0ffd> > -- Channel SIP/5002-006e 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Moving from res_sip to pjsip and simple bridge
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/5349-0072 answered Local/SIP-5710@MemberConnector-0031;2 -- Local/SIP-5710@MemberConnector-0031;1 connected line has changed. Saving it until answer for SIP/5002-006e -- Local/SIP-5710@MemberConnector-0031;1 answered SIP/5002-006e -- Channel SIP/5349-0072 joined 'simple_bridge' basic-bridge -- Channel Local/SIP-5710@MemberConnector-0031;2 joined 'simple_bridge' basic-bridge -- Stopped music on hold on SIP/5002-006e -- Channel Local/SIP-5710@MemberConnector-0031;1 joined 'simple_bridge' basic-bridge <55d1ecc5-b251-4e8b-b0f3-ea08254c0ffd> -- Channel SIP/5002-006e 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 signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
