Sorry about the trouble. Unsubscribed that user from the mailing lists.
Matthew Fredrickson
On Fri, Aug 7, 2020 at 9:20 PM Elizabeth wrote:
>
> I'm online on this site!
> So contact me in my profile:
> here
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> -- Bandwi
I'm online on this site! So contact me in my profile:
galleries.daswanitailors.com here
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Check out the new Asterisk community forum at: https://communit
John,
What you see it's how it should be.
The wait for admin means that all users join the conference room but the
conference is not started and they all should hear MOH.
When the admin will join then the conference will start and all will hear
the admin (or all others if they are not muted)
To all:
No matter what I try, I cannot get the system to wait for the admin to join. It
just dumps users into the bridge directly.
I do not have a pin for users, does that matter?
What am I missing?
Another issue the absolute timeout is not working ? ... have recordings that
last for over 24 h
Thank you Jöran
I also figured out my problem with the caller id name/number. In case anyone
else encounters the caller id name issue, replace the spaces in the name with
control sequence for a space %20
From: asterisk-users On Behalf Of
Jöran Vinzens
Sent: Friday, August 7, 2020 12:10 PM
To
Hi Dan,
as far as PPI and PAI Header, we use the channel Vars in order to do that.
In Latest Asterisk you can set Channel vars within the create command in
the Body. Just set the PJSIP(add,PAI) as you would do it in Dialplan.
https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/
An additional follow-up question, if I need to set the P-Asserted-Identity on
the create (originate), is there a way to do this with ARI?
From: asterisk-users On Behalf Of Dan
Cropp
Sent: Friday, August 7, 2020 11:51 AM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] With ARI,
I'm trying to transition from AMI to ARI.
Running into a small hiccup when I try to create (originate a call) with the
caller id name and number
I can pass the Name and Number if the name has no spaces in it and it shows up
in my PhonerLite application.
curl -v -u asterisk:asterisk -X POST
ht
Hi Carlos
Le 07/08/2020 à 06:33, Carlos Chavez a écrit :
I am having a strange problem with a new provider. We already
have a couple SIP trunks working fine. We are trying a new provider
but we are having one way audio problems with outgoing calls. Incoming
calls do have two way audio, o
