Hi,
We are searching for a pool of test numbers to call from Asterisk, record voice
and test it with our non-intrusive voice quality testing software (NIQA). The
problem is that we could find some test numbers, but our customer would like to
have a global pool of test numbers, so that we can
dear flaviormiranda
thanks for your kind of help. I want to know this part( mfcr2 ) what does
it mean?
signalling=mfcr2
mfcr2_variant=br
mfcr2_get_ani_first=no
mfcr2_max_ani=20
mfcr2_max_dnis=4
mfcr2_category=national_subscriber
mfcr2_logdir=span1
mfcr2_call_files=yes
mfcr2_logging=all
HI group,
this is very strange problem with me when i received a call from Germany i
am able to receive call on my PRI line
everything is fine User connected with IVRS and user trying to enter a
extension number like *1660976
*call goes to users company extension starting with *16.*
is this
On 10/26/2010 06:30 PM, Andrew Latham wrote:
snom phones can do http digest authentication...
I think this digest authentication is for accessing the phone's web
interface, not for contacting a provisioning server
Jonas.
--
On Tue, 2010-10-26 at 17:31 +0200, Jonas Kellens wrote:
Hello,
has anyone experience with auto provisioning IP-phones on different
locations through a central public provisioning server ? You use http
or https ?
Is there a danger that one uses a different MAC-address in the
provisioning
On Tue, Oct 26, 2010 at 11:31 AM, Jonas Kellens jonas.kell...@telenet.bewrote:
Hello,
has anyone experience with auto provisioning IP-phones on different
locations through a central public provisioning server ? You use http or
https ?
Is there a danger that one uses a different
On Wed, Oct 27, 2010 at 4:04 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
On Tue, 2010-10-26 at 17:31 +0200, Jonas Kellens wrote:
Hello,
has anyone experience with auto provisioning IP-phones on different
locations through a central public provisioning server ? You use http
or https ?
On Wed, 2010-10-27 at 04:10 -0400, Steve Totaro wrote:
On Wed, Oct 27, 2010 at 4:04 AM, Ishfaq Malik i...@pack-net.co.uk
wrote:
On Tue, 2010-10-26 at 17:31 +0200, Jonas Kellens wrote:
Hello,
has anyone experience with auto provisioning
On 10/27/2010 10:06 AM, Steve Totaro wrote:
On Tue, Oct 26, 2010 at 11:31 AM, Jonas Kellens
jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote:
Hello,
has anyone experience with auto provisioning IP-phones on
different locations through a central public provisioning
Hi,
On 10/27/2010 05:55 AM, GBR Icasiano, Ryan A. wrote:
anyone???
regards,
RYAN ICASIANO
Hi,
I changed my sip.conf and added call-limit. At first I thought it works ok,
since i tried calling a cellphone that is currently busy(phone answers 1st
softphone, then another softphone calls
{\rtf1\ansi\ansicpg1252\fromhtml1 \fbidis \deff0{\fonttbl
{\f0\fswiss\fcharset0 Arial;}
{\f1\fmodern Courier New;}
{\f2\fnil\fcharset2 Symbol;}
{\f3\fmodern\fcharset0 Courier New;}}
{\colortbl\red0\green0\blue0;\red0\green0\blue255;}
\uc1\pard\plain\deftab360 \f0\fs24
{\*\htmltag19 html
Hi List,
Can anyone please tell me how to use the phoneprov.conf to provision my
client's atas. I read the file but dont know how to actually use it.
--
Best Regards
Rizwan Qureshi
--
_
-- Bandwidth and Colocation Provided by
Hi,
On Tue, Oct 26, 2010 at 05:31:00PM +0200, Jonas Kellens wrote:
Hello,
has anyone experience with auto provisioning IP-phones on different
locations through a central public provisioning server ? You use http or
https ?
What is it exactly that you want to guarantee?
Authenticating
Hi!
We've experienced asterisk has gone without any message, it wasn't any
segfault, anything in asterisk messages log that says about shutting
down.
Asterisk process has just diapered.
Has anybody got similar problem?
Asterisk is version 1.4.29-1 from digium repository.
--
You can read some here http://www.asterisk.org/astdocs/node272.html or
here
http://etel.wiki.oreilly.com/wiki/index.php/Dynamic_Phone_Provisioning_with_res_phoneprov_and_TFTP
There will be more on this topic in the coming months...
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about
Jonas
A quick look at the snom wiki will tell you that I am right...
On 10/26/2010 06:30 PM, Andrew Latham wrote:
snom phones can do http digest authentication...
I think this digest authentication is for accessing the phone's web
interface, not for contacting a provisioning server
On 27/10/2010 12:59, Krzysztof Urbaniak wrote:
Hi!
We've experienced asterisk has gone without any message, it wasn't any
segfault, anything in asterisk messages log that says about shutting
down.
How do you launch asterisk ? did you try without 'safe_asterisk' or
anything like it,
just
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nile Kaledon
Sent: Wednesday, October 27, 2010 4:15 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dial Plan Conf
Jigar,
You should use Read()
On Wed, 27 Oct 2010, Nile Kaledon wrote:
You should use Read() instead of Background() component.
We conf file weenies call them applications.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Wednesday, October 27, 2010 4:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial Plan Conf
Hi!
I've turned off t.38 and all of the codecs except ulaw; I still have the
same problems. SOMETIMES it works. Other times, the sniffer clearly
shows that the media simply isn't being sent. NOTHING is being sent.
Anything else I should check?
Look at the firewalls and possible SIP ALGs
There are NO ACL's in place, either at the network level, or application
level. We have a public address, so as far as I know, there are no forwarding
rules in place.
On Wednesday 27 October 2010 4:04:16 pm Philipp von Klitzing wrote:
Hi!
I've turned off t.38 and all of the codecs except
I have recently updated from Centos/*1.2 to Ubuntu Server and FreePBX
2.8.0.2.
We have an Astribank with 4 T1 ports and 16 FXS ports. After updating, we
had it working for a while with one NT PRI and one TE PRI and, in the
process of trying to configure another PRI, I ran into a couple
Do you have canreinvite=yes anywhere? If yes, try setting it to no. Also
pasting your sip.conf here would be helpful.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-10-27 6:16 PM, Mike Diehl mdi...@diehlnet.com wrote:
There are NO ACL's in place, either at the
On Tuesday, October 26, 2010 01:16:29 pm Stephen Reese wrote:
http://messinet.com/trac/wiki/AsteriskGVGateway (AGI script)
Is your .agi and .git the same script? I do not have a git client on
this host to see for myself.
I keep the AGI in Git as a version control system. But, you can view
Hi,
Thanks for your reply. I'm calling a normal phone using the DIAL cmd. Here is
my sample dial command:
exten =s,4,Dial(SIP/xxx${extensi...@media_gateway,10,t)
but when I use:
exten =s,5,NoOp(SIP/xxx${extensi...@media_gateway has state ${DIALSTATUS})
I hear a busy tone, after the 10 sec.
On Wed, 27 Oct 2010, Nile Kaledon wrote:
You should use Read() instead of Background() component.
On Wed, 27 Oct 2010, Steve Edwards wrote:
We conf file weenies call them applications.
On Wed, 27 Oct 2010, Danny Nicholas wrote:
Is that like a Perl Weenie?
Yes, and you can proudly wear as
Hello community,
I've been running Asterisk on an embedded device for about six months, and
my operation has been largely trouble-free. I'm hoping I could get some help
with a minor problem:
Every week or three, my PBX gets stuck in a state where it can receive
calls, but it becomes completely
Can they be used from any unsecured access point (eg they have a browser
to enter in a password etc) or can you only use them from home AP's etc.
Cheers,
Dean
-Original Message-
From: asterisk-biz-boun...@lists.digium.com [mailto:asterisk-biz-
boun...@lists.digium.com] On Behalf
Hello all,I'm working with one of our offices (that is moving soon) and they're being offered ISDN-10/20/30 services from their TELCO. I'm wondering what kind of interface card I will need (I prefer using Digium's cards). Are the TE121/122/ or TE212/220 series cards compatible with this kind of
On 10/27/2010 09:21 PM, Cassius Smith wrote:
I'm working with one of our offices (that is moving soon) and they're
being offered ISDN-10/20/30 services from their TELCO. I'm wondering
what kind of interface card I will need (I prefer using Digium's cards).
Are the TE121/122/ or TE212/220
On Tue, Oct 26, 2010 at 8:26 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
I'm going to try and look at this during Astricon :)
Ok, just uploaded a new patch on
https://issues.asterisk.org/view.php?id=18202 Let me know if it
worked.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Hi all,
Are there anyone use ss7_lib or ss7_channel in production ?.
What about its quality and reliablity ?.
Can an Asterisk servce with ss7_lib or ss7_channel can processs 480 conccurent
call (8 E1 line) ?
Many thanks,
Giang
--
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