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...strations, though, have been tweaked so that they can exist in realtime but require a reload to be picked up. Those are the two that come to mind. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5kaWdpdW0uY29t&r=YmFzZQ%3D%3D & http://spamaway.npgco.com/canit/urlproxy.php?q=aHR0cDovL3d3dy5hc3Rlcmlzay5vcmc%3D&r=YmFzZQ%3D%3D -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by htt...

Hi, The asterisk wiki page says: "Sorcery.conf allows you to try to configure other PJSIP objects such as transport using realtime and it currently won't stop you from doing so. However, some of these object types should not be used with realtime and this can lead to errant behavior." Which objects and is this still true in 1.13.1 ? Thanks, Antonio. PS:

Hello, I am currently trying to set up pjsip realtime and would like to have outbound-publish, inbound-publication, and asterisk-publication sorcery object types in ODBC realtime. Is that currently supported? I know that some object types are known working and others are not. I was curious what the status of those objects are. Thanks! Matt Hoskins | NPG Corp | Systems Architect

...possible with recent changes that have been done. I personally have not tried it, and it would still require a reload to pick up changes regardless. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: http://spamaway.npgco.com/canit/urlproxy.php?_q=aHR0cDovL3d3dy5kaWdpdW0uY2 9t&_r=YmFzZQ%3D%3D & http://spamaway.npgco.com/canit/urlproxy.php?_q=aHR0cDovL3d3dy5hc3Rlcmlzay 5vcmc%3D&_r=YmFzZQ%3D%3D -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by...

...s scrubbed... > URL: < > http://lists.digium.com/pipermail/asterisk-users/attachments/20141030/8cbc01fe/attachment-0001.html > > > > ------------------------------ > > Message: 10 > Date: Thu, 30 Oct 2014 17:01:42 -0500 (CDT) > From: Matt Hoskins <matt.hoskins at npgco.com> > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> > Subject: [asterisk-users] MWI publish VIA pjsip for non sip channels > Message-ID: <9a3428ea.000019bc.00000006 at IT10015vm.npgco.com>...

Matt Hoskins wrote: > Good Morning, > > After further investigation, I found that the res_pjsip_publish_asterisk > module does not use the realtime sorcery wizard, but instead only reads > from the configuration files. I've been able to patch the module, using > the logic from the other modules to learn how to make the sorery > configuration read from the other sorcery

...possible with recent changes that have been done. I personally have not tried it, and it would still require a reload to pick up changes regardless. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: http://spamaway.npgco.com/canit/urlproxy.php?_q=aHR0cDovL3d3dy5kaWdpdW0uY2 9t&_r=YmFzZQ%3D%3D & http://spamaway.npgco.com/canit/urlproxy.php?_q=aHR0cDovL3d3dy5hc3Rlcmlzay 5vcmc%3D&_r=YmFzZQ%3D%3D

Hello, I'm currently evaluating asterisk 13 (Currently on 11). We're testing the migration from SIP to PJSIP. Is there a way to alias the SIP channeltype to PJSIP when exlusively using pjsip? Matt Hoskins | NPG Corp | Systems Architect 816.749.2815 (Internal: ext. 10015) -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an asterisk 11.4.0 server with two interfaces, eth0 and eth1. Eth0 has a default gateway on it, eth1 is connected the subnet that has my phones registered. I'd like to use the multicastRTP driver to do paging. However, when a phone dials an extension with multicastRTP, the multicast stream goes to the primary interface (eth0) and it really needs to go to eth1. Is there a

Before I go down a rabbit hole, does the mwi publish/subscription work for non SIP phones? For instance, I have a single voicemail server, connected to multiple asterisk boxes via SIP. On each of those servers, there are a mix of SIP and SCCP phones attached. Currently, I'm using res_xmpp to distribute mwi from the voicemail server to the endpoint servers. Would this type of setup work