On Wed, Sep 7, 2022 at 11:35 AM Antony Stone <
antony.st...@asterisk.open.source.it> wrote:
> On Wednesday 07 September 2022 at 15:32:50, Thomas Ray wrote:
>
> > From https://wiki.asterisk.org/wiki/display/AST/Channels
> >
> > "The primary exception is with Local Channels. In the case of local
>
From: asterisk-users on behalf of
"Joshua C. Colp"
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Wednesday, September 7, 2022 at 10:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Channel names with semicolons
On Wednesday 07 September 2022 at 15:32:50, Thomas Ray wrote:
> From https://wiki.asterisk.org/wiki/display/AST/Channels
>
> "The primary exception is with Local Channels. In the case of local
> channels, you'll typically have two local channel legs, one that is
> treated as outbound and the
Hello - I am using asterisk 18.14.0
I think multicast uses codec g711 pcmu
is there any way to change or set the codec I want to use - like g722 ?
How would I do that?
Thanks
Jerry
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On Wednesday 07 September 2022 at 11:44:54, Antony Stone wrote:
> Hi.
This is a follow-up to an email I posted earlier today to the list, although I
haven't seen it come back yet. If it's under moderation for some reason, I
hope some kindly admin will release it :)
> I'm trying to deal with
On Wed, Sep 7, 2022 at 11:17 AM Antony Stone <
antony.st...@asterisk.open.source.it> wrote:
> On Wednesday 07 September 2022 at 11:44:54, Antony Stone wrote:
>
> > Hi.
>
> This is a follow-up to an email I posted earlier today to the list,
> although I
> haven't seen it come back yet. If it's
Hi.
I'm trying to deal with a problem regarding putting a call on hold and then
later resuming it. I am using chan_sip throughout, and Asterisk 16.
I have two scenarios:
First (works):
1. An inbound call arrives, the dialplan does not Answer() it.
2. The dialplan performs a Dial() to an
On Wednesday 07 September 2022 at 15:21:59, Joshua C. Colp wrote:
> On Wed, Sep 7, 2022 at 11:17 AM Antony Stone wrote:
> >
> > This is a follow-up to an email I posted earlier today to the list,
> There's nothing in the moderator queue that I can see.
Thanks, sent again, and immediately
From https://wiki.asterisk.org/wiki/display/AST/Channels
"The primary exception is with Local Channels. In the case of local channels,
you'll typically have two local channel legs, one that is treated as outbound
and the other as inbound. In this case both are really inside Asterisk, but one
On 9/7/2022 1:06 PM, Jerry Geis wrote:
> I think multicast uses codec g711 pcmu
> is there any way to change or set the codec I want to use - like g722 ?
The CHANGES entry for the MulticastRTP channel type[1] has some info on
this. Basically you would add the 'c' option to your dial string:
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