On 11/01/2019 09:19, Administrator TOOTAI wrote:
Le 11/01/2019 à 10:12, Neil Youngman a écrit :
A while back, I posted about detecting when a call was picked up by a
fax machine. It was suggested that having a "fax" extension and
"faxdetect=yes" would cause it to jump to the "fax" extension.
Hello,
I've been asked if it is possible or not to set several (10 or so) SIP
trunks between two boxes, one beeing an Avaya IPBX, the other being an
Asterisk 13 or 16 box (with either chan_sip or pjsip).
The reason behind this question come from billing requirements.
I'm not convinced yet
Le 11/01/2019 à 10:12, Neil Youngman a écrit :
A while back, I posted about detecting when a call was picked up by a
fax machine. It was suggested that having a "fax" extension and
"faxdetect=yes" would cause it to jump to the "fax" extension. This was
not something I could get to work.
I
A while back, I posted about detecting when a call was picked up by a
fax machine. It was suggested that having a "fax" extension and
"faxdetect=yes" would cause it to jump to the "fax" extension. This was
not something I could get to work.
I have now created a very simple example. In
On 11/01/2019 10:08, Administrator TOOTAI wrote:
I don't understand your goal. You want to send or receive fax?
I'm attempting to deliver a voice message. If the message has been
mistakenly sent to a fax I want to detect that it's a fax and report that.
Neil Youngman
Neil Youngman
On 11/01/2019 10:04, Floimair Florian wrote:
I would guess from your explanation that the "outgoing" call somehow ends up in
your Asterisk machine again, either at the voicetest or fax extension.
You don't answer it in either of the extensions.
That's what TOOTAI meant.
No, it's not looped
Le 11/01/2019 à 10:23, Neil Youngman a écrit :
On 11/01/2019 09:19, Administrator TOOTAI wrote:
Le 11/01/2019 à 10:12, Neil Youngman a écrit :
A while back, I posted about detecting when a call was picked up by
a fax machine. It was suggested that having a "fax" extension and
I would guess from your explanation that the "outgoing" call somehow ends up in
your Asterisk machine again, either at the voicetest or fax extension.
You don't answer it in either of the extensions.
That's what TOOTAI meant.
If this is done in another extension, than this part of the Dialplan
Le 11/01/2019 à 16:47, Matt Riddell a écrit :
Hiya,
When I hang up on a call to my stasis app I’m getting multiple channelDestroyed
events for the same channel:
app.js:985:13) Channel was destroyed: 1547220509.77
app.js:1029:17) This was a customer
app.js:1030:17) Checking if this was a
Did something recently change for the chan_sip bindport setting?
I know I had this working with the previous version of asterisk. Can't
remember if it was an earlier 16.x version or 13.x
I was running chan_sip (binding to port 5061) and PJSIP using the default port
of 5060.
I recently
Please disregard.
I just found my problem. In my sip.conf, I had an include statement for
another file at the top. The included file also had a general section. Once I
deleted that general section and reloaded everything worked as expected, now
using port 5061.
From: asterisk-users On
> On Jan 11, 2019, at 11:14, Jean Aunis wrote:
>
> Le 11/01/2019 à 16:47, Matt Riddell a écrit :
>> Hiya,
>>
>> When I hang up on a call to my stasis app I’m getting multiple
>> channelDestroyed events for the same channel:
>
> It may happen if several applications subscribed to the channel.
>Hi,
>On the other side.. There is a specific note regarding CDR behavior changes
>from asterisk 12 onwards. So going from 1.8 to 13 means it applies to you.
>Have a look at:
>https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12
>And
Hiya,
When I hang up on a call to my stasis app I’m getting multiple channelDestroyed
events for the same channel:
app.js:985:13) Channel was destroyed: 1547220509.77
app.js:1029:17) This was a customer
app.js:1030:17) Checking if this was a customer talking to an agent
app.js:1043:21) Customer
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