A site question: which of the following RFC would describe as-feature-event
?
[1] https://www.iana.org/assignments/sip-events/sip-events.xhtml
Le mer. 1 mars 2017 à 21:03, Trey Hilyard a écrit :
> Is there any "easy" way to add a custom subscribe handler? I have a set of
> users with Polycom
> Am 15.01.2019 um 15:23 schrieb Doug Lytle :
>
> Hi all,
>
> When moving from a self compiled Asterisk 13.23.1 to Asterisk 13.24.0, has
> resulted in a MWI clearing delay of around 5 minutes.
>
> After listening to a voicemail and deleting it, the Polycom VVX 601's MWI
> light is left on
>>> https://github.com/astlinux-project/astlinux/commit/3bfd9f0400e990a42e1317f4aa2bad51a3ef9f17
>>> I am using "mailboxes=##@default" and had the issue as well (before).
>>> Michael
Thanks Michael!
I'll try that patch later on today. I'm not using the mailboxes=##, but will
try the patch
Hello,
There is question that bounces in my mind for quite a long time.
Today, I dare to ask it here:
how do you package and use your custom asterisk .deb package ?
The background is:
- I'm now a long time Debian user and I learned to appreciate Debian's deb
package benefits specially when
Hi all,
When moving from a self compiled Asterisk 13.23.1 to Asterisk 13.24.0, has
resulted in a MWI clearing delay of around 5 minutes.
After listening to a voicemail and deleting it, the Polycom VVX 601's MWI light
is left on for around five minutes, before clearing.
Installing Asterisk
On Tue, Jan 15, 2019, at 9:29 AM, Olivier wrote:
> A site question: which of the following RFC would describe as-feature-event ?
>
> [1] https://www.iana.org/assignments/sip-events/sip-events.xhtml
If I recall correctly it doesn't have a spec, it's one of the custom things
Broadsoft has done
Hi all,
Is there a way with Polycom phones or alternatives, to configure a specific
SIP server for such as-feature-event or call-info events ?
If positive, maybe a third party SIP server (Kamailio, ...) supporting
those events would allow such implementation.
Looking at Yealink phone Admin
Carlos and Stefan (and other who have helped):
I DON'T HAVE the res_timing_timerfd.so file. Can I build it? Recompiling
Asterisk is unrealistic in my position but I wonder if I can build the one
module. Here's what I do have:
apbx:~ $ locate *res_timing_timerfd*
>>> I'll try that patch later on today. I'm not using the mailboxes=##, but
>>> will try the patch just the same.
Patch applied and fixed my problem,
Doug
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Subject: Re: [asterisk-users] Various extensions ring once and goto
voicemail - Thomas Peters
>Carlos and Stefan (and other who have helped):
>I DON'T HAVE the res_timing_timerfd.so file. Can I build it? Recompiling
>Asterisk is unrealistic in my position but I wonder if I can
>>> Carlos and Stefan (and other who have helped):
Thomas,
You stated that your virtual environment was Oracle, would that equate to
VirtualBox?
Doug
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On Tue, 2019-01-15 at 09:00 -0700, John Kiniston wrote:
> How is your endpoint currently configured in asterisk?
It's configured as a chan_sip peer.
> Have you tried
> rtp_symmetric to see if the endpoint sends audio to asterisk if
> asterisk
> can send audio back to the client?
That would
Hi Guys
I've run into a weird problem on Asterisk 13. Again something that worked fine
on 1.8 but is now broken on Asterisk 13.
I have an extension 3015. I'm trying to originate a recording playback call on
it via AMI by sending
Action: Originate
ActionID: test
Channel: SIP/3015
Exten:
On Tue, Jan 15, 2019, at 12:18 PM, Brian J. Murrell wrote:
> On Tue, 2019-01-15 at 09:00 -0700, John Kiniston wrote:
> > How is your endpoint currently configured in asterisk?
>
> It's configured as a chan_sip peer.
>
> > Have you tried
> > rtp_symmetric to see if the endpoint sends audio to
Hello,
I've just gone through the process of cross-compiling Asterisk 16 for
ARM. I thought it would be as easy as calling the "./configure" script
with the appropriate "host" parameter, but it turned out to be more
complicated. I'm wondering whether I did something wrong, or if there
are
How is your endpoint currently configured in asterisk? Have you tried
rtp_symmetric to see if the endpoint sends audio to asterisk if asterisk
can send audio back to the client?
Alternatively if your SIP Proxy is also a Media proxy you could set the
media_address on the endpoint to be your proxy
On Tue, 2019-01-15 at 12:01 -0500, Joshua C. Colp wrote:
>
> The chan_sip module has this implemented under the "nat" option using
> "comedia" as I recall.
Yeah. The help for which reads:
Send media to the port Asterisk received it from regardless
of where the SDP says to send it.
> It causes
This is going to be a bit of an odd situation, but perhaps might become
more and more common (as mobile phone SIP clients utilize PUSH proxies
instead of the battery draining direct registering with SIP servers).
I have a SIP client which can be on the same RFC-1918 LAN as my
Asterisk server.
On Tue, Jan 15, 2019, at 1:17 PM, Brian J. Murrell wrote:
> On Tue, 2019-01-15 at 12:01 -0500, Joshua C. Colp wrote:
> >
> > The chan_sip module has this implemented under the "nat" option using
> > "comedia" as I recall.
>
> Yeah. The help for which reads:
>
> Send media to the port Asterisk
In article <018201d4acef$898a4b10$9c9ee130$@verishare.co.za>,
Stefan Viljoen wrote:
> Hi Guys
>
> I've run into a weird problem on Asterisk 13. Again something that worked
> fine on 1.8 but is now broken on Asterisk 13.
>
> I have an extension 3015. I'm trying to originate a recording playback
Hi Tony
Ok, got this solved.
I discovered my AMI message was corrupt due to a bug in our third party dialer
app we wrote ourselves...!
E. g. this worked on Asterisk 1.8:
ActionID=12edad43-e817-427b-aa21-31a9659f86e1
=Originate
=SIP/local/3035@local
=
=local
=1
=3035
=recordinglisten
I'm running Fedora 29. asterisk starts with a systemd service at boot.
On any reboot I get a LOT of :
[Jan 15 09:30:26] ERROR[1162]: netsock2.c:541 ast_sockaddr_hash: Unknown
address family '0'.
[Jan 15 09:30:35] ERROR[1161]: netsock2.c:541 ast_sockaddr_hash: Unknown
address family '0'.
[Jan
From https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces:
res_timing_dahdi uses timing mechanisms provided by DAHDI. This method
of timing was previously the only means by which Asterisk could receive
timing. It has the benefit of being efficient, and if a system is
already going to
Actually, I was wrong about that. We no longer use OVM. It's actually Citrix
Xencenter 7.6
Thomas M. Peters | Sr. Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org
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