Nice, Do you have the code up on GitHub? I'd love to see it.
What's the source of the data? Something API driven I hope?
Have you thought about implementing your project via curl instead of
func_odbc?
On Wed, May 27, 2020, 8:52 PM Saint Michael wrote:
> In a few weeks, no SIP call is going to
Yes, this means that a provider which only provides IP-access (for example a
broadband operator), ergo, when it doesn’t terminate a call, but where the call
terminates directly at a enterprise, does not need to force the end customer to
implement call verification in their PBX.
Basically, if
On Thu, 28 May 2020, Saint Michael wrote:
My company is one if the six service providers approved.
Which part of 'Non-Commercial' do you not understand? The topic may be of
general interest. Hawking your wares is not.
--
Thanks in advance,
>
> My company is one if the six service providers approved. We are not ready
> yet, probbably next week, since the certificate needs to be issued by the
> Certification Authority. As I said, we are the ONLY provider that you may
> use with Asterisk remotely, via UnixODBC. The rest of the other
On 2020-05-28 11:10, Doug Lytle wrote:
But if you've already got the caller on the phone, then you might consider
the CONNECTEDLINE function in Asterisk...
>
> And that we don't.
>
> It's the third party that would like the notification the the destination
> phone is currently busy
>>> But if you've already got the caller on the phone, then you might consider
>>> the CONNECTEDLINE function in Asterisk...
And that we don't.
It's the third party that would like the notification the the destination phone
is currently busy with another call. CONNECTEDLINE only functions
On 2020-05-28 10:15, Doug Lytle wrote:
> Everybody,
>
> I've had a request from my manager that I figure out how to get our Asterisk
> 13.x system using chan_sip to be able to display on the Polycom VVX series
> phone display (firmware 5.9.5), when an extension is called and the person on
>
Everybody,
I've had a request from my manager that I figure out how to get our Asterisk
13.x system using chan_sip to be able to display on the Polycom VVX series
phone display (firmware 5.9.5), when an extension is called and the person on
the other end is on the phone.
He said, "Our old
Thank you Joshua.
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Wednesday, May 27, 2020 5:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Is it possible to have a single AMI originate
ring multiple contacts?
On Wed, May 27, 2020 at
