Michele Pinassi
2018-Feb-22 13:18 UTC
[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/5349-00000072 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 - Universit? 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 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 181 bytes Desc: OpenPGP digital signature URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180222/181f7ecd/attachment.pgp>
Joshua Colp
2018-Feb-22 18:08 UTC
[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/5349-00000072 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