To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP/SDP for MulticastRTP page
Matthew Murphy wrote:
> Hi everyone,
>
>
> I am sending out a multicast page using the following in my dialplan:
>
>
> Page(MulticastRTP/linksys/
Hi everyone,
I am sending out a multicast page using the following in my dialplan:
Page(MulticastRTP/linksys/${MULTICAST_IP_ADDR}:6061/${MULTICAST_IP_ADDR}:6061,q)
Everything works great, but I had a question about SIP and SDP:
Should I be seeing a SIP/SDP message from the asterisk server
Hi everyone,
I upgraded from Asterisk 13.5.0 to 13.7.0 and I am having database connection
problems. I am doing Asterisk realtime with PJSIP 2.4.5 and it works perfectly
in 13.5.0. But now I am losing my database connection (running on a virtual
box) and I am stuck!
I spent all day
com> on behalf of A J Stiles
<asterisk_l...@earthshod.co.uk>
Sent: Tuesday, January 19, 2016 7:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Segmentation Fault Asterisk 13.7.0-rc2
(libmysqlclient?)
On Monday 18 Jan 2016, Matthew Murphy
Hi everyone,
I am getting a segmentation fault (seems to occur randomly) using Asterisk
13.7.0-rc2 with PJProject 2.4.5. It appears to be something that libmysqlclient
is complaining about when doing a query in ps_endpoint_id_ips. We are using
Asterisk Realtime. This also seems to occur in
Joshua Colp
<jc...@digium.com>
Sent: Wednesday, January 13, 2016 9:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] "pjsip show endpoints" returns "No Objects Found"
in 13.7.0-rc2
Matthew Murphy wrote:
> Hi everyone,
>
Hi everyone,
I have just upgraded to Asterisk 13.7.0-rc2 and noticed that when I type "pjsip
show endpoints" at the CLI, I get "No Objects Found".
However, if I request information on a specific endpoint, (for example: "pjsip
show endpoint 101") then I get all of the information for that
ion.
On 1/10/2015 1:51 AM, Matthew Murphy
wrote:
Greetings everyone,
I was wondering if there was a way to change the codec that
Asterisk uses when streaming via MulticastRTP. Or perhaps a
way to transcode the multic
Greetings everyone,
I was wondering if there was a way to change the codec that Asterisk uses when
streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream.
In the CLI, when I have a multicast stream in progress, I am typing 'core show
channel MulticastRTP/0x7f7' to
nicely.
Typical PICNIC error: Problem In Chair, Not In Computer.
Thanks for the help!
Date: Tue, 25 Aug 2015 15:00:09 -0300
From: jc...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Changing volume via dialplan
Matthew Murphy wrote:
Greetings everyone,
Kia
Greetings everyone,
I am attempting to adjust the volume of a call using Set(VOLUME) in my
extensions.conf file. I am finding that Set(VOLUME(TX)=x) and Set(VOLUME(RX)=y)
have no discernable effect on my endpoints (Snom 300 IP phones). I have tried
setting x and y to -30, -10, -3, -2, -1, 0, 1,
11 matches
Mail list logo
