I upgraded from a perfectly working 1.6.2 asterisk installation
(including fax via app_fax_digium) to 1.8.0 this evening.
All my custom modules (including swift thanks darren!) are working
fine except for fax.
When a caller connects, asterisk switches to the fax context and hangs
up the call.
Hi,
I've now set up Asterisk to interface with our current Samsung iDCS 100 PBX via
an 8SLI analogue extension card in the Samsung and an Openvox A400P04 4-FXO
card in the Asterisk box. It all works in that I can place calls in both
directions from the office Samsung extensions and Asterisk
Sure thing! Bug #18302 has been opened
(https://issues.asterisk.org/view.php?id=18302).
Brett Woollum
br...@woollum.com
- Original Message -
From: Sherwood McGowan sherwood.mcgo...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
Ronny Adsetts wrote:
Hi,
I've now set up Asterisk to interface with our current Samsung iDCS 100 PBX
via an 8SLI analogue extension card in the Samsung and an Openvox A400P04
4-FXO card in the Asterisk box. It all works in that I can place calls in
both directions from the office
John Novack said at 13/11/2010 12:58:
Ronny Adsetts wrote:
[...]
The problem I'm trying to solve at the moment is getting caller ID
info passed over to the SIP phones when calls are placed. The
caller ID is coming through as 'asterisk' which I assume is the
default if nothing is present. So
Hi Everyone,
I don't have much experience with eSXI. I can really use some advise on how
to run it without any trouble with Asterisk on CentOS VMs.
First of all, is it a good option to run multiple hosted Asterisk instances
on a VMware eSXI? or would you rather prefer something like Xen,
On 11/13/2010 4:36 AM, Jeremy Kister wrote:
When a caller connects, asterisk switches to the fax context and hangs
up the call.
I was wrong, asterisk does not even switch to the fax extension-
i added a noop, and it's not making it:
exten = fax,1,NoOp( in fax extension )
exten =
At 05:56 AM 11/13/2010, you wrote:
John Novack said at 13/11/2010 12:58:
Ronny Adsetts wrote:
[...]
The problem I'm trying to solve at the moment is getting caller ID
info passed over to the SIP phones when calls are placed. The
exten = s,1,Verbose(1,Samsung 209 ${CALLERID(all)})
And I
Hi,
I'm using qualify= on my asterisk server that provides outgoing pstn calls to a
few companies.
I've got one client in particular that has their own asterisk server which is
connected to my server.
This client seems to be having a nat issue. It's not a connectivity issue as
i've tried
Hi!
I want my PBX to be reachable at my ekiga.net account. It seems I am
registered:
vajna2*CLI sip show registry
HostUsername Refresh
StateReg.Time
ekiga.net:5060 magwas 585
Registered Sat, 13 Nov 2010
This does sound like something that should stay on Asterisk-users.
On Sat, Nov 13, 2010 at 3:36 AM, Jeremy Kister asterisk...@jeremykister.com
wrote:
I upgraded from a perfectly working 1.6.2 asterisk installation
(including fax via app_fax_digium) to 1.8.0 this evening.
So you made sure to
On 11/12/10 23:14, Joseph wrote:
After upgrade to mysql 5.1.51 my asterisk-stat-v2 is not displaying correctly.
Does anybody have a similar problem? Is it due to mysql-5.1.51 or the problem
is with new glibc-2.11.2 ?
--
Joseph
I see that asterisk-stat-v2 was replaced with CDR-Stats
The
Between 10:00AM and 10:00PM CST on Saturday, November 13, the services
below will experience extended outages as the servers that host them are
upgraded and reconfigured:
downloads.digium.com
downloads.asterisk.org
bamboo.asterisk.org
packages.asterisk.org
svn.digium.com
svn.asterisk.org
Ronny Adsetts wrote:
John Novack said at 13/11/2010 12:58:
Ronny Adsetts wrote:
[...]
The problem I'm trying to solve at the moment is getting caller ID
info passed over to the SIP phones when calls are placed. The
caller ID is coming through as 'asterisk' which I assume
Here is a very very basic config. But, not working (:
I simply want to dial the DID that is registered with the SIP provider.
then, as you can see the call should dial the 703111 number
Hints please?
sip.conf
;register = 908366554:396...@carrier.jazzey.com
register =
Try changing this line:
exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks))
To:
exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks))
Sent from my iPhone
On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote:
Here is a very very basic config. But, not
Hello,
I have a few questions regarding Asterisk 1.8.0. If you can answer a
question, please do so.
Is Asterisk 1.8.0 stable enough for production environments?
Is it possible (and if yes what is the best option) to use CDR MySQL with
Asterisk 1.8.0? With 1.6.x we use the add-on package for
Hi Brett,
It did not work.
I will try other ideas.
SIP or Dial plan problem?
registeration?
On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote:
Try changing this line:
exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks))
To:
exten =
From: Mark Scholten m...@streamservice.nl
Hello,
I have a few questions regarding Asterisk 1.8.0. If you can answer a
question, please do so.
Is Asterisk 1.8.0 stable enough for production environments?
It appars to be so far we are testing and
What is the error message?
Sent from my iPhone
On Nov 13, 2010, at 6:28 PM, Thomas Perron thomas.per...@gmail.com wrote:
Hi Brett,
It did not work.
I will try other ideas.
SIP or Dial plan problem?
registeration?
On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote:
i am running 1.4.37 and am hosted on Rackspace.
I feel like a took a step back by using the Cloud server service since
I am having a little trouble proving that my basic configuration is
working.
Nevertheless, I want to upgrade to 1.8.
I use Centos 5.5
Anyone know of a good link that can help
How do I see the error message?
the phone call seemed to get through but I did not see anything on my
1.4 console.
i used 1.6.x before. having trouble with this for some reason. older stuff.
i have one session open at the prompt but nothing shows up.
On Sat, Nov 13, 2010 at 9:53 PM, Brett
You get into asterisk by saying asterisk -r. You then up the verbosity by
saying core set verbose 3 or some such number. You the call your number and
you should see the steps of your dialplan execute.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Nov 13, 2010, at
Jim,
Thanks. But, no joy.
I set to 3, then 5.
I don't think I am getting registered somewhere.
The console shows nothing.
The call to the DID drops after 5 seconds or so.
It does not ring.
I know. Basic stuff. I really think the version of this code is not
robust enough.
My sip.conf and
On Sat, Nov 13, 2010 at 8:00 PM, Thomas Perron thomas.per...@gmail.comwrote:
i am running 1.4.37 and am hosted on Rackspace.
I feel like a took a step back by using the Cloud server service since
I am having a little trouble proving that my basic configuration is
working.
Nevertheless, I
11 nov 2010 kl. 23.25 skrev Baha @ SH:
Hello
How can I run the sip service on asterisk on another port beside 5080?
I mean asterisk will still take sip requests on port:5080 and another custom
port, lets say port:6080
For UDP, we only have one port. You have to select.
/O
--
10 nov 2010 kl. 21.48 skrev Hans Witvliet:
On Wed, 2010-11-10 at 08:38 +0100, Olle E. Johansson wrote:
6 nov 2010 kl. 15.30 skrev Hans Witvliet:
Hi all,
As stated in the subject, slightly off-topic, as it is not directly a
Asterisk issue, but more SIP in general
Because security in
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