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
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
Anyway Thank you guys. ;-)
On Wed, Feb 19, 2014 at 12:25 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
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
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,
On Tue, Feb 18, 2014 at 3:40 AM, Torbjörn Abrahamsson
torbjorn.abrahams...@gmail.com 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
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 rnew...@digium.com wrote:
On Tue, Feb 18, 2014 at 3:40 AM, Torbjörn Abrahamsson
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
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 -
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
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
On Wed, Feb 19, 2014 at 12:12 PM, Torbjörn Abrahamsson
torbjorn.abrahams...@gmail.com 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
On Tue, Feb 18, 2014 at 2:13 PM, Steve Edwards
asterisk@sedwards.com 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
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
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