Hi all
Using asterisk 16.25.0 (I know it's a bit old)
I'm trying to parse the realtime queue_log and I realized that not every call
has a ENTERQUEUE event.
I see that if there's an internal call to an extension that has a dialplan
forward to a queue (no call is placed to the extension before
Hi all
Using asterisk 16.25
I was trying to stop Mixmonitor using features. The code is executed but I
realized that I was executing StopMixmonitor from another channel so I opted to
use AMI.
When I call MixMonitor I store the channel name in a var and then I use
StopMixmonitor from AMI
El Thu, 24 Nov 2022 15:08:12 +0500
Tahir Almas Dhesi escribió:
> What cps and maximum concurrent calls on following hardware configuration
> with freeswitch possible ?
>
> I need approximate speculation not exact or real time results
>
> Intel E-2286 G 2 x 960 SSD , 6 cores / 12 threads
El Wed, 27 Apr 2022 12:27:03 +1200
David Cunningham escribió:
> Hello,
>
> Does anyone know of a way to have a call go to a particular context when a
> 302 Moved is received in response to an invite? This is with chan_sip. We
> tried setting __TRANSFER_CONTEXT but it didn't seem to have any
Hi all
I'm experiencing issues while trying to restrict ExtenSpy to a single user's
channels.
I'm trying this:
same => n,ExtenSpy(${EXTEN:3},bde(${EXTEN:3}))
and this too
same => n,ExtenSpy(${EXTEN:3},bde(SIP/${EXTEN:3}))
Being EXTEN:3 == "ext20012104" in this scenario
And in the logs
El Sat, 10 Jul 2021 23:02:10 +0200
Antony Stone escribió:
> On Saturday 10 July 2021 at 22:57:09, Eric Wieling wrote:
>
> > > On Sat, Jul 10, 2021 at 2:39 PM Alexander Perkins wrote:
> > >
> > > Hi All. We have a provider that requires us to SOURCE the SIP
> > > connection on TCP 5061.
El Tue, 3 Nov 2020 16:20:22 +0100
Olivier escribió:
> Hello,
>
> How would you test how a PJSIP-powered Asterisk 13 instance resist to
> hostile REGISTRATION attempts ?
>
> Would you use SIPp ? Any example scenario ?
> Would you go with an alternative tool ? Which one would you pick ?
>
>
El Tue, 27 Oct 2020 12:52:47 -0400
Dovid Bender escribió:
> Hi,
>
> Sorry in advance that I am emailing the users list and not the biz list I
> think I will find my target audience here. We are looking to hire a
> consultant to help us figure out an issue. We are having what seems are
> "random
El Thu, 9 Jul 2020 10:37:52 +0200
Antony Stone escribió:
> On Thursday 09 July 2020 at 00:50:28, Jon Bonilla (Manwe) wrote:
>
> > DO you know odbc redis drivers? It would be nice to store cdrs ans other
> > stuff in redis without patching asterisk
>
> A quick Google
El Wed, 8 Jul 2020 12:23:42 -0700
John Kiniston escribió:
> Dovid, You could use func_odb + a ODBC Redis driver to keep from having to
> shell out.
>
>
DO you know odbc redis drivers? It would be nice to store cdrs ans other stuff
in redis without patching asterisk
--
PekePBX, the
El Tue, 9 Jun 2020 09:46:32 -0600
George Joseph escribió:
Hi George
>
>
> >
> > If transcoding is enabled Would it be possible to do the same but handle a
> > 488
> > back from Bob and failover to another INVITE with Bob's allow list to
> > handle
> > transcoding? That way we would always
El Fri, 5 Jun 2020 09:42:33 -0600
George Joseph escribió:
> Greetings All,
>
> We've been working hard on new codec negotiation stuff for Asterisk 18 and
> we've got some stuff to run by you. It's a lot so please read carefully.
Hi Joseph
Thanks for such a detailed mail and involving the
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