"C.Maj" writes:
> Another option for a patch would be to extend the PJSIP_DIAL_CONTACTS
> function with an argument such as 'please' to minimally return the
> endpoint name in a Dialable format when no reachable contacts are found
> eg. "PJSIP/bar" -- instead of the current empty string, which
On 2019-11-26 09:17, Greg Troxel wrote:
> For the second issue, it would be nice if Dial just discarded empty
> destinations, as in
>
> Dial(PJSIP/foo&)
> Dial(PJSIP/foo&/baz)
>
> as would result from the following if there were no bar registrations
>
>
Hi All. I had a situation today where all the dynamic agents became
'available'. This is a backup system we have that our folks call in from
their cell phones and wait for calls while we fix the primary system.
However, when we tested today, they all appeared as 'unavailable' and no
calls that
>> So which option is preferred?
>>
>> A) Have a softphone aor/auth_user/password for a particular human, and
>> expect them to configure it on multiple devices. Do not worry that 1)
>> multiple are registered at once (because that's normal in SIP) and 2)
>> asterisk has no idea which is
Hi Gang
I offer:
g722
g711a
g711a is mandatory. g722 is becoming more and more popular.
Now if a call originates from a device which support g722 and ends on a
device which does not. I see that asterisk is transcoding between g722
and g711a despite both ends supporting g711a.
Google tells me,
I'm no expert on the user side of things, but I would prefer option A. Of
course, this is completely your preference. Asterisk will allow either
option, so you have some flexibility there. One of the advantages of option
A is that you can have multiple devices (like you mentioned) that can all
be
On 11/26/19 12:31 AM, Jonathan H wrote:
Yes, I know I post similar back in January, but there was no response
back then and I was hoping things might have changed :)
I'm using IBM's Watson for voicemail transcriptions, they allow 500
minutes per month for speech to text on the Free/Lite plan.
Hi Gang
If anyone else stumbles over the same Problem.
This is how I solved it for now:
On the IC Trunk:
trust_id_inbound=no => Makes sure the CallerID is taken from the From Header.
trust_id_outbound=yes => Does nothing useful, maybe a bug?
send_pai=no
On the incoming call, you have to pull
