Tilghman,
Tx, That was the solution.
Kind Regards,
Arjan Kroon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: dinsdag 29 januari 2008 16:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote:
Hi all
I have a small problem here. I asked this question on another asterisk
mailing list, but nobody seemed to be able to help me there.
We are running
* Asterisk 1.4.17
* Libpri 1.4.3
* Zaptel 1.4.8
on a 1.6 dual
Hi,
I have installed GNU gatekeeper. Then I am trying to install asterisk addons. I
gave make and then make clean. I worked properly. Then I gave make install. It
gave following error.
make[1]: Entering directory `/usr/src/asterisk/asterisk-addons/asterisk-ooh323c'
cp
On Wed, Jan 30, 2008 at 01:38:48PM +0100, Joost Kuif | Mobillion wrote:
Hi,
I can't read a environment variable in a asterisk dialplan.
When logged in as user root on the system an 'echo $HOSTNAME' gives the
hostame of the machine.
Asterisk (1.4) is started from the same console.
I
Hi,
I can't read a environment variable in a asterisk dialplan.
When logged in as user root on the system an 'echo $HOSTNAME' gives the
hostame of the machine.
Asterisk (1.4) is started from the same console.
I try to read it like this:
exten = s,n,NoOp(host=${ENV(HOSTNAME)})
Does anyone
You're using the variables wrong.
This is what you could do is either:
exten = s,1,NoOp(${ENV(HOSTNAME)})
or
In globals section
;; Defing hostname
host=${ENV(HOSTNAME)})
In you dailplan section
exten = s,1,NoOp(${host})
You will manage,
Greets
Joris
On Jan 29, 2008 8:36 PM, Steve Totaro [EMAIL PROTECTED] wrote:
On Jan 29, 2008 5:55 PM, Alex Balashov [EMAIL PROTECTED] wrote:
Franklin,
Because ChanSpy() is a passive monitor, there is nothing about the
implementation that would cause Asterisk to shunt the speech back to
itself.
Maybe...
exten = s,n,NoOp(SET(host=${ENV(HOSTNAME))})
?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joost Kuif
| Mobillion
Sent: 30 January 2008 12:39
To: asterisk-users@lists.digium.com
Subject:
This pointed me into the right direction, thanks Tzafrir!
i added a export HOSTNAME=$HOSTNAME into my .bash_profile
Grtz,
Joost
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Tzafrir Cohen
Verzonden: Wednesday, January 30, 2008 1:58 PM
Aan:
On Wed, 30 Jan 2008, Joost Kuif | Mobillion wrote:
This pointed me into the right direction, thanks Tzafrir!
i added a export HOSTNAME=$HOSTNAME into my .bash_profile
Grtz,
Joost
I explicitly pass it to Asterisk in this snippet from my
/etc/init.d/asterisk file.
Steve Totaro wrote:
I was under the impression that only ABE supports Dialogic boards. I
thought I saw that in passing so I could be totally wrong.
There was talk but ABE has never supported Dialogic cards. If anyone would be
interested, I would recommend expressing it to [EMAIL PROTECTED]
Douglas Garstang wrote:
Does anyone know the cause of these BAD BAD BAD messages?
I think I lost all my calls when it happened too. We have nagios running
against IAX and nagios reports that IAX is down. It would seem that the
entire application locks up when this happens and calls are
Hi,
is it possible to use asterisk to serve two SIP-domains with different
users?
It does work to define two domains with 'domain=' in sip.conf, but that
allows all sip-users to register with both domains.
I want to define users for a one domain only and not allow them to use
the second.
Lets
Octavio Ruiz wrote:
Allen Casteran wrote:
Anyone know what the POE draw is for the Aastra 480i phones?
We have switches that will do 15 watts on 12 ports but only do 7.7 watts on all
24 ports.
A Cisco 3560 switch will do 15.6 watts on all 24 ports.
Just trying to find out if we need that much
Thanks to both of you for your input. I'll be in touch off list Steve.
-Franklin
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 30, 2008 8:00:48 AM (GMT-0500)
Hi,
I have one problem, i´ve a trunk sip Asterisk--- Cisco 2600. Call
inbound work very good, but call outbound don´t work. Call progress but no
audio. Canreinvite=no , no Nat, No problem Codec.
Any idea???
Thanks in advance,
D
___
--
Hi,
I am using Debian OS kernel 2.6.22-3-amd64
and zaptel driver 1.4 with ztdummy module for meetme application.
I use meetme with SIP channels.
I have such problem that when one connects to the conference voice is cut.
Each voice sequence is disturbed.
Does any one have similar issue and
Hello,
we are migrating an Asterisk install from Trixbox/Asterisk 1.2 to Elastix
0.92 with asterisk-1.4.17 on CentOS 5. I need to migrate an funtion that
consults a remote sybase database, using ODBC and freetds. On the new
server I am able to connect to the database using isql without problems.
30 jan 2008 kl. 15.44 skrev Bjoern Haje:
Hi,
is it possible to use asterisk to serve two SIP-domains with different
users?
It does work to define two domains with 'domain=' in sip.conf, but
that
allows all sip-users to register with both domains.
I want to define users for a one domain
See below:
Dirk Enrique Seiffert wrote:
Hello,
we are migrating an Asterisk install from Trixbox/Asterisk 1.2 to Elastix
0.92 with asterisk-1.4.17 on CentOS 5. I need to migrate an funtion that
consults a remote sybase database, using ODBC and freetds. On the new
server I am able to connect
On Jan 30, 2008 3:43 PM, Russell Bryant [EMAIL PROTECTED] wrote:
There have been 380 fixes made to Asterisk 1.2 since 1.2.14, 61 of which were
made to chan_sip. :)
Did the one that gives the message Yikes, we should NEVER BE
HERE! get swatted? :)
Tomasz Zieleniewski wrote:
I am using Debian OS kernel 2.6.22-3-amd64
and zaptel driver 1.4 with ztdummy module for meetme application.
I use meetme with SIP channels.
I have such problem that when one connects to the conference voice is
cut.
Each voice sequence is disturbed.
Does any
ztdummy can give you issues as a timing device. Any way you could try using a
Digium card just as a timing device to see if this helps?
- Original Message -
From: Tomasz Zieleniewski [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
XXX 4. res_config_odbc
XXX 13. res_odbc
ODBC Resource
Depends on: unixodbc(E), ltdl(E)
unixODBC and unixODBC-devel are installed.
You also need ltdl (whatever that is) ;)
I guess this
libtool-ltdl-1.5.22-6.1
... which is installed.
Thanks
Enrique
Thanks for any hint to get my
Is there any way to have Asterisk call an extension in dial plan instead of
original extension after timeout?
Like extension A puts the caller in parking lot, he leaves the phone and
forgets about it, instead of having that phone rings after timeout, have a
group of phones rings.
Sorry if this was repeated, but I think the list is acting up and not
accepting some emails so I wanted to resend it just in case.
On Jan 29, 2008 10:15 AM, arkda [EMAIL PROTECTED] wrote:
Recently with Asterisk 1.4.17 I've been running into some stability
issues. I started looking through my
Dirk Enrique Seiffert wrote:
I guess this
libtool-ltdl-1.5.22-6.1
... which is installed.
Thanks
Enrique
I believe you're looking for libtool-ltdl-dev(el)
--
Jason Parker
Digium
___
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On 1/27/08, Michelle Dupuis [EMAIL PROTECTED] wrote:
The Aastra's also have a range of interested firmware bugs that
support/development just can't seem to fix. Do a search for aastra
hang/lockup and you will find what I mean.
Have you ever achieved those hangup cases?
Which firmware/model
Dears,
After weeks trying to contact support of my telecom about 'Seize Ack'
because that is not returned, was a lock for make calls on my E1s.
Now I receive back de Ack and get ready to make calls, but the technical
support reports to me that my attempts to call do not send any digits to
the
Running asterisk 1.2.23 on a meetme
two polycom SIP phones. When we both hung up the
meetme was still active.
Is there something special needed to destroy the static conference?
Jerry
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Asterisk 1.4.18-rc3 is now available. The important bug fixes that made it
into
this RC are a couple of crash fixes for ChanSpy/MixMonitor. A few other less
severe bug fixes made it in, as well.
This release candidate is published for anyone that is interested in helping to
test it for a
Well, that's simple, the telco is not getting any digits because YOU
are not sending any digits!
From the logs, I see you are dialing like this:
Dial(UniCall/g1|300|)
Where is the number you want to reach?
I'd expect to see
Dial(Unicall/g1/1234567890|300)
To reach number 1234567890
- Moisés
My apologies for the OT request, I think more people here may have what I am
looking for than on the commercial list (which I did post to also).
I'm looking for a used 2 FXS port pci card, don't care who makes it as long
as it works. Oh and I have a limited budget...;)
If you have anything you
At the end of the day SIP calles keep stuck in asterisk, is there any
way to prevent this or debug this?
The sip calls which get stuck all are calles on a krik IP600v3 dect
gateway,
I cant tell if they originate of the ip600v3, probably this are calls TO
the IP600v3
10.0.0.71240
Dirk Enrique Seiffert wrote:
I guess this
libtool-ltdl-1.5.22-6.1
... which is installed.
Thanks
Enrique
I believe you're looking for libtool-ltdl-dev(el)
Thansk a lot, - this made the difference!!! 2 days of unhappy hacking
found an end!!!
--
Jason Parker
Digium
Hi List;
The output of cat /proc/cpuinfo giving a [Intel (R)
Pentium (R) D] so what is the g729 version I have to
download to work with my machine?
Any help?
Regards
Bilal
Looking for last minute
Franklin wrote:
ztdummy can give you issues as a timing device.
Yes and no. See below
Any way you could try using a Digium card just
as a timing device to see if this helps?
Tomasz wrote:
I am using Debian OS kernel 2.6.22-3-amd64
and zaptel driver 1.4 with ztdummy module for meetme
Greetings,
I have an issue with the length of time that passes from when someone hits the
transfer soft key on a Cisco 7940, after doing an attended transfer, and when
the recipient’s connects with the transferred call. It appears to be around 6
seconds. Is there a .conf in Asterisk where
At 07:09 AM 1/30/2008, you wrote:
We have had 5 of the Netgear switches in production for almost a
year now, each powering 5 Aastra 480i phones without any issues whatsoever.
I have one of the 8 port 4 with POE Netgear boxes powering 3 480i-CT
phones for 2 years or so with zero probalems.
Ira
Try the CT-G1000 from cyber-telecom.net it is 39.99 GBP atm.
Sam
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos Rojas
Sent: Wednesday, January 30, 2008 10:35 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk gateway
Hello everybody
On Thu, Jan 31, 2008 at 05:59:03AM +0800, Sam Tam wrote:
Try the CT-G1000 from cyber-telecom.net it is 39.99 GBP atm.
And how do they compare to others?
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
Actualy Aastra phone are limited to control only 50 BLF, snom360 can
handle in our site about 110, and only seems a bit busy time to time.
adrià vidal
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asterisk-users mailing list
On Thu, Jan 31, 2008 at 05:59:03AM +0800, Sam Tam wrote:
Try the CT-G1000 from cyber-telecom.net it is 39.99 GBP atm.
err biz again ...
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac
Man I am a little embarrassed now...
Actually dial plans and PBX rules is where I have less knowledge of
everything that involves the asterisk, because of this I am using freePBX
and this was my problem. I make the setup for outbound trunk to UniCall
using the freePBX and in this case has a
Hi everyone,
My manager interface seems to be producing wrong CallerIDs when
internal extensions call each other. Can anyone see anything wrong in
the configuration snippets pasted below? The following instance has
extension 101 call 103. The phone does show the right caller ID, but
notice that
remove the brackets around
(Devraj Mukherjee 101)
Regards
Kev
Devraj Mukherjee wrote:
Hi everyone,
My manager interface seems to be producing wrong CallerIDs when
internal extensions call each other. Can anyone see anything wrong in
the configuration snippets pasted below? The following
Hi,
I am new to asterisk configuration.
I want to get called number in features.conf.
I am defining a feature in features.conf and that feature got executed on
pressing a particular DTMF key sequence.
As I want to execute my own application on pressing that key which will use
called number.
Ok, I have spent all night trying to figure this out, and hopefully
somebody has a similar experience.
I have a very basic asterisk config. Sample configs, with the only
addition being by SIP phone, and my incoming voip. Last week I got
everything setup, calls were working, etc.,.
I cam
You might need Voicemailmain([EMAIL PROTECTED])
PaulH
On Thu, 2008-01-31 at 00:30 -0500, John Von Essen wrote:
Ok, I have spent all night trying to figure this out, and hopefully
somebody has a similar experience.
I have a very basic asterisk config. Sample configs, with the only
Can you get some verbose output from your console/logs? It may be more
obvious once you see what Asterisk is attempting to do when this
extension is dialed.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Von
Essen
Sent: Wednesday, January 30, 2008
John Von Essen wrote:
Ok, I have spent all night trying to figure this out, and hopefully
somebody has a similar experience.
I have a very basic asterisk config. Sample configs, with the only
addition being by SIP phone, and my incoming voip. Last week I got
everything setup, calls were
Franklin Webb wrote:
Thanks to both of you for your input. I'll be in touch off list Steve.
-Franklin
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 30,
Tried it, but no change.
A few updates. Even though I dont hear anything, if I hit a keys on the
phone and then hang up, message log says:
[Jan 30 21:26:57] WARNING[7917] app_voicemail.c: Unable to read password
I enabled logging of everything, and the below is the snippet for when
my
How about your sip.conf for your extensions?
Example:
[6001]
host=dynamic
type=friend
disallow=all
allow=ulaw
I usually don't see this (I'm more production and haven't done heavy
debug for a long time):
[Jan 30 21:26:50] DEBUG[7917] channel.c: Set channel SIP/6001-081de7a8
to write format gsm
Digium has a compatibility list of servers, however, it has not been updated
since 2006. One of the servers on the list has since been taken out of
production by Dell. Here are the remaining servers on the list: HP Proliant
DL360IBM x206IBM x346
Does anyone has a most recent list and I will be
You need $dnis.
On Jan 30, 2008, at 11:08 PM, Prashant Sharma
[EMAIL PROTECTED] wrote:
Hi,
I am new to asterisk configuration.
I want to get called number in features.conf.
I am defining a feature in features.conf and that feature got
executed on pressing a particular DTMF key sequence.
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