I'm curious as to what versions of everything you are using. Reason
being this line -- DAHDI/i1/00256312261627-1 is proceeding passing
it to SIP/5000-.
It states DAHDI/i1/00256312261627-1... and I don't recall seeing that
before (my 2.4.0 says -- DAHDI/1-1 is proceeding passing
Hello list,
I'm having some troubles with DTMF tones. When pressing numbers on a Snom
phone, the DTMF-signal takes too long.
I had this problem after upgrading Asterisk. What version of Asterisk are you
currently using?
Has the Snow worked fine before? Or have you always had the
Hi Andrew,
I am using current versions of software, find below versions:
1.) asterisk
voice:~# asterisk -V
Asterisk 1.8.2.3
2.)dahdi
*CLI dahdi show version
DAHDI Version: 2.4.0 Echo Canceller: MG2
3.) lipri
*CLI pri show version
libpri version: 1.4.11.5
I've already tried to call over each
Hi Danny,
Thanks again for your help !
Exten = 4001,1,Dial(DAHDI/g1/5551212)
Or
Exten = 4001,1,Dial(SIP/5551...@myprovider.com)
It looks like I'd be using Dial(SIP/...) as for other numbers I have a
macro such as this:
[macro-dialSIP]
exten = s,1,Dial(SIP/${ARG1})
exten =
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Axelle
Sent: Monday, February 21, 2011 11:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Assigning an extension to
Thanks Faisal, in fact I made a test that confirmed that in realtime
asterisk doesn’t supported static peers, like you told me.
Do you know if newer versions of asterisk, like 1.8, have this issue already
solved?
Regards,
Ricardo.
On Wed, Feb 16, 2011 at 6:26 PM, Faisal Hanif
Hello everybody!
I get this message when making outbound calls:
[Feb
21 14:24:46] WARNING[25204]: chan_sip.c:3621 __sip_autodestruct:
Autodestruct on dialog 'ee162385cac5cc9c@10.1.1.13' with owner in place
(Method: BYE)
All inbound calls are fine.
In other SIP users everything seems fine
On Mon, Feb 21, 2011 at 11:33 AM, Danny Nicholas da...@debsinc.com wrote:
[Feb 21 17:53:06] WARNING[26195]: chan_sip.c:2921 create_addr: No such
host:
001
[Feb 21 17:53:06] WARNING[26195]: app_dial.c:1202 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
On Mon, Feb 21, 2011 at 12:37 PM, Warren Selby wcse...@selbytech.comwrote:
Then you need to include the [roaming-ext] context in whatever context your
phones dial from. The basic idea behind this is that you need to store the
extension where your roamer is currently sitting in your DB, which
We are running Asterisk version 1.4.23-1, libpri-1.4.9 and
zaptel-1.4.12.1 and two Digium TE220Ps. Debugs are set to 10.
We have a T1 PRI connected to the telco. Over the last 4-5 days, we
have getting Yellow/Red alarms coming from the T1 PRI. The other two
ports in use are connected to
Dean,
what's your zaptel Zapata config_
regards
Juan.
Linux User #441131
On Mon, Feb 21, 2011 at 1:44 PM, Dean Hoover kb7...@gmail.com wrote:
We are running Asterisk version 1.4.23-1, libpri-1.4.9 and
zaptel-1.4.12.1 and two Digium TE220Ps. Debugs are set to 10.
We have a T1 PRI
Hi all,
Sorry for being a little off topic, but I just need some tips on some good
provider that offers free calls to the US. I have tried out one called
Whistlephone, but I am not able to receive calls with it and when I use the
follow me feature it still rings here. So any other I should try?
Here you go:
/etc/zaptel.conf:
loadzone = us
defaultzone=us
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
#Added 2nd 2xT1 card
span=3,0,0,d4,ami
em=49-72
span=4,0,0,d4,ami
fxoks=73-96
---
/etc/asterisk/zapata.conf:
I don't see any problem.. but, i don't see the 2nd SPAN @ zaptel:
*yellow alarm on span 2*
regards.
Juan.
Linux User #441131
On Mon, Feb 21, 2011 at 2:11 PM, Dean Hoover kb7...@gmail.com wrote:
Here you go:
/etc/zaptel.conf:
loadzone = us
defaultzone=us
span=1,0,0,esf,b8zs
Your message was:
Here you go:
/etc/zaptel.conf:
loadzone = us
defaultzone=us
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
#Added 2nd 2xT1 card
span=3,0,0,d4,ami
em=49-72
span=4,0,0,d4,ami
fxoks=73-96
Ooops, my bad I Did not read the zaptel config file correctly, my apologize.
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
*span=2,1,0,esf,b8zs
bchan=25-47
dchan=48*
Juan.
Linux User #441131
On Mon, Feb 21, 2011 at 2:34 PM, Juan David Diaz juanch...@gmail.comwrote:
Your message was:
have you check the PRI crossover cable?
Juan.
Linux User #441131
On Mon, Feb 21, 2011 at 2:35 PM, Juan David Diaz juanch...@gmail.comwrote:
Ooops, my bad I Did not read the zaptel config file correctly,
my apologize.
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
*span=2,1,0,esf,b8zs
That is certainly an option, but my query was more on the side of is
it possible to trace/log the ESF frames. From what I have read, there
are signals going back and forth from a lower level than the
D-channel, and in ESF there are codes that can be sent to the other
side to reset the frame. If
This doesn't represent the 2nd span?
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
Dean
On Mon, Feb 21, 2011 at 1:18 PM, Juan David Diaz juanch...@gmail.com wrote:
I don't see any problem.. but, i don't see the 2nd SPAN @ zaptel:
yellow alarm on span 2
regards.
Juan.
Linux User #441131
You should try paying for the call, and then you will be able to get good
service
CS
On February 21, 2011 at 2:07 PM Christian christia...@runbox.com wrote:
Hi all,
Sorry for being a little off topic, but I just need some tips on some good
provider that offers free calls to the US. I have
Greetings,
If you have an account on issues.asterisk.org and just received an email about
having an account created on a JIRA server, please ignore and accept our
apologies for the accidental message. We were doing some test migrations of
data from issues.asterisk.org into a newer issue
Asterisk Project Security Advisory - AST-2011-002
Product Asterisk
Summary Multiple array overflow and crash vulnerabilities in
UDPTL code
http://i-wikisport.com/product.php?page=32a
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Hello
On a brand new CentOS 5.5 host, I wanted to install Dahdi before
installing Asterisk 1.4.x.
So I untarred dahdi-linux-complete-2.4.0+2.4.0.tar.gz, followed by
make, make install, make config.
Next, I created the following files from scratch:
/etc/modprobe.d/dahdi.conf:
Hi everyone,
does anyone know is AstriEurope coference is still on ?
I saw some notification about cancelation on website, but cant find it
anymore.
Regards,
Albert
--
_
-- Bandwidth and Colocation Provided by
I just installed an FXS module onto a 4 channel tdm thats about 5
years old and it wont work. Running dmesg I can see the following
error:
Zapata Telephony Interface Registered on major 196
Freshmaker version: 71
Freshmaker passed register test
!!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000
AstLinux v0.7.6
Asterisk v1.4.39.1
I have found that in some recent version of Asterisk v1.4, NVFaxDetect has
started to cause segfaults on my system. I am using the following code in
extensions.conf:
[incoming]
exten = 7057974880,1,Answer
exten = 7057974880,n,NVFaxDetect(4|t)
exten =
On 2/21/11 4:52 PM, Gilles wrote:
Hello
On a brand new CentOS 5.5 host, I wanted to install Dahdi before
installing Asterisk 1.4.x.
So I untarred dahdi-linux-complete-2.4.0+2.4.0.tar.gz, followed by
make, make install, make config.
[snip]
Here's what tail /var/log/messages shows:
On 2/21/11 4:46 PM, C F wrote:
I just installed an FXS module onto a 4 channel tdm thats about 5
years old and it wont work. Running dmesg I can see the following
error:
Zapata Telephony Interface Registered on major 196
Freshmaker version: 71
Freshmaker passed register test
!!!
On Monday 21 February 2011 21:11:28 Shamus Rask wrote:
3. Is it true that Digium is sidelineing IAX2 and only focusing on
SIP? Should I be looking to migrate to SIP trunks instead?
Is it true that space aliens stole your brain and replaced it with a head
of cabbage?
--
Tilghman
--
On Mon, Feb 21, 2011 at 11:56 PM, Albert alber...@wp.pl wrote:
does anyone know is AstriEurope coference is still on ?
http://www.astrieurop.com/fr/cloture.php
Cancelled.
Hello,
It is with regret that we announce you the cancellation of the
AstriEurop exhibition on May, 3rd and 4th 2011 in
Are there other European Asterisk conferences?
Thanks,
Sevana Oy
Vendor of Asterisk VQM
- Original Message -
From: randulo rand...@randulo.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 22, 2011 9:56 AM
Subject:
32 matches
Mail list logo
