Yes, you can do this. You should point the trunks to the right context
and done.
Op 30-10-2012 8:15, Darin Iv schreef:
Hi all,
I need to configure DIDs for different companies and they should reach
on different extension with different context. Cant we have same
extension in different
Elliot,
I am installing by default wireshark/thark on my asterisk machines.
This allows to do protocol traces from the linux commandline and to
store the traces into a file. You can find more information at
www.wireshark.org
Henk
Elliot Murdock schreef:
Hello,
When debugging SIP
Daniel,
Have you thought about using CURL from the Dialplan?
Henk
Daniel Isenmann schreef:
Hi,
is it possible to configure a TCP trigger to a predefined address and
port on a incoming call request?
Some background:
For example “Client 1” tries to call “Client 2”, “Client 1” is sending
Try to find a pattern. Looks that you are able to reproduce the
problem. You mention after 4 minutes. Is this also the case for
internal calls? If so then I would say that the E1 is ok. If not then
I would step more into E1 related issues. Have you looked at the
ethernet cards.
I have been playing with this some time ago. We used the so called mode
code integration. This worked fine. It works simular as described for
other Avaya Product.
http://www.voip-info.org/wiki/view/Avaya+or+Lucent+Magix+Voicemail+Integration
Henk
BJ Weschke schreef:
I've just
Doug,
Have you checked the feature access code that is defined in the
definity. That is the code that needs to be dialed. I always checked
the codes from a definity phone to make sure that I was using the right
codes.
Henk
Doug Lytle schreef:
Henk Dick wrot
I have been playing
Did you install mpg123? You can check from linux prompt by just typing
mpg123
Bhrugu Mehta schreef:
no , not at all, there is no need to install sound card in asteirsk system.
I am using asterisk server without soundcard.
so there may be antoher problem may in configurtion of zapata or
You can use sip show peers. If an IP address is shown then the user will be
available.
Henk
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I would check:
Cat /proc/zaptel/
To make shure that the cards are activated in the order that you programmed
them.
Henk
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Backup
e-mail
Sent: maandag 30 april 2007 13:26
To: asterisk-users@lists.digium.com
Which codec do you plan to use?
Henk
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of umar tarar
Sent: woensdag 7 februari 2007 20:05
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] CPU motherboard for 100+ simultaneouse calls
I think that you are loading the drivers in the wrong order. You can change
the order of loading are first define the E1 followed by the TDM400
Hope this helps,
Henk
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lincoln
Zuljewic Silva
Sent: woensdag 22
I would suggest the following
- remove the drivers
- load them manually (zaptel, wcte11xp, wctdm)
Run:
Zttools - should show unconfigured cards.
Take:
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
fxsks=32-35
loadzone = us
defaultzone=us
run:
ztcfg -vv
See what it is
This is what I am using:
exten = o,1,Answer()
exten = o,2,GoTo(default,3000,1)
exten = o,3,Hangup()
Hope this helps,
Henk
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony Davis
Sent: maandag 24 juli 2006 18:20
To: Asterisk
Users Mailing List
()
exten = o,2,GoTo(default,3000,1)
exten = o,3,Hangup()
Hope this helps,
Henk
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Anthony
Davis
*Sent:* maandag 24 juli 2006 18:20
If you do not use USB then I would suggest to disable this in the bios.
Henk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Harper
Sent: donderdag 29 juni 2006 14:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE
Did you try to manually to
change the parameters of the phone? When you power the phone up
then are you able to enter manually the parameter when you hit *. I
am using a 4610 with Release 2.2 but I am not using the capability to upload
the settings from the server.
Henk
From
I dont think that this will work. It is Avayas own implementation of H323
and not standard H323. When you want to use the phone you have to migrate
them to SIP.
Henk
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos Rojas
Sent
Have a look at the attached
link.
http://support.avaya.com/japple/css/japple?temp.documentID=283920temp.productID=107755temp.bucketID=108025PAGE=Document
Henk
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gabriel Rosca
Sent: dinsdag 6 juni 2006 21:15
Create the 2 extensions in /etc/asterisk/extension.conf
exten = 8,1,Answer()
.
Script 1
.
exten = 9,1,Answer()
.
Script 2
.
Make sure that the channel where the calls come in route the call to the
context where you defined the scripts.
Hope this helps,
Henk
Try to do ztcfg s before
you run ztcfg -vv
Henk
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Louis curty
Sent: woensdag 31 mei 2006 12:52
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] Need
help with Junghanns
This should do the job
exten = _91NXXNXX,2,Dial(${SIPtrunk}/${EXTEN:1},55,o)
Henk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez
Sent: dinsdag 30 mei 2006 22:01
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
I think that you are sending an outgoing caller id that is not part of the
DID range. Most operators do not allow this.
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1013' ]
Are you using caller id 1013 ?
Change it to a number that is part of your trunks.
Henk
-Original Message
Marty,
Just remove the options for each technology.
Dial(SIP/2005IAX/2010,25,tr)
This should do the job
Henk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph
Sent: zaterdag 28 januari 2006 9:27
To: Asterisk Users Mailing List - Non
I think that you still should be able to use the voice mail system of our
service provider. It will detect that all 3 lines are busy and reroute the
call.
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Brad Stockdale
Verzonden: woensdag 23 februari 2005 20:31
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