Matthew,
I don't think I've been as clear as I'd like.
I've got some fax-connected TA's that make outbound calls. The dial
plan routes those calls to an AGI script that captures the fax image,
the destination phone number, and creates a call file to deliver the
image to the destination.
The fir
On Tue, Feb 18, 2014 at 2:13 PM, Steve Edwards
wrote:
> On Mon, 17 Feb 2014, Mike Diehl wrote:
>
>> Is there something I need to do in order to get the h extension to get
>> called?
>
>
> Would the 'g' dial() option help?
>
> "Proceed with dialplan execution at the current extension if the destina
On Wed, Feb 19, 2014 at 12:12 PM, Torbjörn Abrahamsson
wrote:
> Thank you very much. I will try this! It seems to be what I'm looking for.
>
> I'm in most cases working with 1.2 asterisks, so I'm not up to date on newer
> features. My current project however needed a newer version. I tried some
Thank you very much. I will try this! It seems to be what I'm looking for.
I'm in most cases working with 1.2 asterisks, so I'm not up to date on newer
features. My current project however needed a newer version. I tried some
googleing, but I did not find these variables.
Thanks,
Torbjörn Abra
Hi list,
I have a fresh install of Asterisk 12.0.0 and I'm going to use it only
as a client. I'm trying to SIP REGISTER with a remote SIP provider.
The situation is that Asterisk is running in a VMware VM with a RFC IP
address (192.168.1.2). The provider of the VM performs static NAT from
th
A reboot of the system after hours appears to have solved the issue.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg
Sent: Wednesday, February 19, 2014 11:15 AM
To: Asterisk Users Mailing List - Non-Commerc
Eric!
The pcap trace seems to contain only idle data, and there is nothing unusual.
As far as I know, PRI always uses the static TEI value of 0, as there is only a single
"terminal equipment (TE). The TEI assignment procedure is only relevant if there is a BRI
connection on a so called "S0
Actually SIPFROMDOMAIN was documented here:
https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
, but SIPFROMUSER was not. They are now both there! :)
On Wed, Feb 19, 2014 at 9:08 AM, Rusty Newton wrote:
> On Tue, Feb 18, 2014 at 3:40 AM, Torbjörn Abrahamsson
> wrote:
>> I have
On Tue, Feb 18, 2014 at 3:40 AM, Torbjörn Abrahamsson
wrote:
> I have a problem where I would like to be able to send an arbitrary SIP
> domain when sending a call to a registered friend. By default the from
> domain is set to the IP of the Asterisk server, but I would like to set it
> to somethin
It seems a layer 2 problem, should check about references point (network or
terminal equipment), it assume your Asterisk box is connected to ISDN PSTN
provided, just in case check at you side if all related configuration files
are configured as signalling=pri_cpe (Card config, wan_cfg, chan_da
Anyway Thank you guys. ;-)
On Wed, Feb 19, 2014 at 12:25 PM, A J Stiles
wrote:
> On Wednesday 19 Feb 2014, Gholamreza Sabery wrote:
> > Hello, a few days ago I sent a question:
> >
> >
> http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html
> >
> > but no one answered me! I
On Wednesday 19 Feb 2014, Gholamreza Sabery wrote:
> Hello, a few days ago I sent a question:
>
> http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html
>
> but no one answered me! I just want to know is it possible or not?
There is a bit of a tendency on this list to ignore
On 19/2/14 4:53 am, Gholamreza Sabery wrote:
Hello, a few days ago I sent a question:
http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html
but no one answered me! I just want to know is it possible or not?
I can't help on the "can Asterisk detect they're behind the same NA
13 matches
Mail list logo
