: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hoskins
Sent: Wednesday, February 18, 2015 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13 - sorcery realtime for pjsip
publish objects
Awesome!
https://issues.asterisk.org/jira/browse/ASTERISK-24811
Matt Hoskins | NPG Corp | Systems Architect
816.749.2815 (Internal: ext. 10015)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of George Joseph
Sent
,asterisk-realtime
ps_inbound_publication = odbc,asterisk-realtime
ps_asterisk_publication = odbc,asterisk-realtime
Matt Hoskins | NPG Corp | Systems Architect
816.749.2815 (Internal: ext. 10015)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users
what the status of those objects are.
Thanks!
Matt Hoskins | NPG Corp | Systems Architect
816.749.2815 (Internal: ext. 10015)
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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
Joshua,
Regarding Outbound Registrations in realtime and a reload. Does it require
a pjsip reload or full asterisk restart?
Antonio Gómez Soto wrote:
Hi,
The asterisk wiki page says:
Sorcery.conf allows you to try to configure other PJSIP objects such
as transport using realtime and it
to distribute mwi
from the voicemail server to the endpoint servers. Would this type of
setup work with PJSIP? The net effect here is that I want to get away
from res_xmpp, if possible.
Matt Hoskins | NPG Corp | Systems Architect
Of course, I left out a detail that may (or may not change) the answer.
I'm using the external chan-sccp-b sccp module, not the chan_skinny
bundled with asterisk.
Matt Hoskins | NPG Corp | Systems Architect
816.749.2815 (Internal: ext. 10015)
-Original Message-
From: asterisk
Awesome - Thanks for the quick replies. I'm sure I could have
tried-and-see but with going from Asterisk 11 to 13, there'd be so many
things changing - it helps to know from the outset.
Thanks again.
Matt Hoskins | NPG Corp | Systems Architect
-Original Message-
From: asterisk
I love the templates and used them extensively in the beginning of my
asterisk journey. But abandoned them once I went realtime. Moving them
to realtime - WOULD BE AWESOME!
Matt Hoskins | NPG Corp | Systems Architect
816.749.2815 (Internal: ext. 10015)
-Original Message-
From
Holy damn - I didn't know you could use templates in extensions! Mind =
Blown.
Matt Hoskins | NPG Corp | Systems Architect
816.749.2815 (Internal: ext. 10015)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry
Brummell
Sent: Thursday, March 13, 2014 1:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
rwhee...@artifact-software.com
Subject: Re: [asterisk-users] Replying to
, the multicast stream goes to
the primary interface (eth0) and it really needs to go to eth1.
Is there a way to specify which interface the rtp is sourced from?
Matt Hoskins | NPG Corp | Systems Architect
816.749.2815
enough
for production use.
Thank you for your time,
Matt Hoskins
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Alright, I'm inspired. I'll give it a shot. Should I use the asterisk
hint system or is line appearance done in the sccp config file seperately?
Do you have a configuration example?
Thanks!
Matt
Sergio Chersovani wrote:
Matt Hoskins ha scritto:
I'm currently using asterisk 1.0.7
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