Displaying 20 results from an estimated 10000 matches similar to: "{s} - extension"
2008 Mar 04
2
Problems configuring Astribank
Hi, all
My Asterisk uses a Digium TE120Pand I would like to add an Astribank
zaptel_hardware sees is, but I cannot get it working
pbx:~# zaptel_hardware
Argument "IRQ" isn't numeric in numeric comparison (<=>) at
/usr/local/share/perl/5.8.8/Zaptel/Span.pm line 114.
usb:005/002 xpp_usb- e4e4:1131 Astribank-8/16 USB-firmware
pci:0000:04:00.0 wcte12xp+
2008 Feb 26
6
[URGENT] Zap channels fail to load
I have spent some time this morning trying to add an Astribank to our
current Asterisk, but it failed, so I just removed the hardware,
restore the config files to the original setup and started asterisk.;
I could see that no Zap channels are started so I did load chan_zap.so:
pbx*CLI> module load chan_zap.so
[Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application:
Already have an
2008 Jan 30
7
Problem with DTMF dialing
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 core, 2GB ram and a digium TDM800P wildcard, hardware echo
cancelation and a quad FXO card.
We have 4 analog lines, one of which is a Cellphone line for least cost
2007 Dec 12
3
Load Balancing over 2 E1 Lines
Hi @ all,
i set a server to a costumer of mine with a TE207P for use with 2 E1 Lines.
I set them together into one group in zaptel/zapata.conf
The point is now, the customer has a free-volumina of 60k minutes per month,
per line.
How can i make a kind of load balancing, that both lines will be trafficed
the same way ?
I read something about DIAL(Zap/r1/.) for using round robin, and
2008 Feb 27
3
About faxes recived through a PRI and passed to a fax machine connected to a FXS port
Hi, all
I want to configure a few FXS ports in an Antribank-16 to be able to
receive faxes sent throught a PRI:
E1 ==>Zap * ==>FXS * ==>Fax machine
My asterisk box has a Digium TE120P (for the PRI).
Versions are *=> 1.4.17 | Zaptel=>1.4.8 | libpri=>1.4.5
The Astribank is not configured yet, because I am a little bit
confused about how to do it.
Let's say I configure
2007 Dec 09
1
Installing/configuring TE120P debian way
Hi all
I use asterisk (1.2 brach) from debian official packages and it works fine.
Now I need to install and configure a Digium TE120P card, but I cannot
find any guide to install it using debian packages.
I would like to know if anyone of you knows about packages that would
include the necessary kernel modules or any other method that won't be
broken when the asterisk packages are updated.
2007 Aug 17
3
Lock extension from asterisk
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi all
I am working in a new set up with Grandstream GXP-2000 handsets. I
like those phone, but they lack a feature I need: the phone cannot be
locked by the user.
What I actually want is a user to be able to avoid someone else making
calls from his phone without giving him access to SIP configuration
access to the phone.
i.e. let say I want user
2007 Aug 18
1
Best way to detect unknown and/or private incoming caller-id?
I am aware of how to match a particular caller-id or a caller-id
pattern and do something with the call like this:
exten => 15554441212/_888NXXXXXX,n,Playback(GoAway)
What I am curious about, is the best way to block unknown, private and
000-000-0000 calls.
I know I can do this for 000-000-0000 calls:
exten => 15554441212/0000000000,n,Playback(GoAway)
Is there a better way to catch
2008 Mar 01
4
Cisco 79xx users/consultants, 7970G color in particular share information
I would like to get in contact with users/consultants who are or have
worked with the Cisco phones and Asterisk to trade information.
Cisco has reluctantly made SIP available on their phones and most of the
information on voip-info and other wiki's appears to be reverse
engineered. There is a wealth of information out there which is
terrific.
I have a client with about 40 phones
2008 Feb 22
5
load balancing SIP extensions
What I would like to do is have two identical *
servers which accept registrations of sip extensions
4000-4999.
If I define a rrDNS or LinuxHA then I should have
load-balanced registrations.
However, say ext. 4001 is registered on *1 and 4002 is
registered on *2, if 4001 tries to call 4002 then I
would like to do something like:
- lookup 4002 on *1, try to establish a call if it's
2006 Nov 24
2
Card don't hangup but Asterisk hangup
Hi ,
I have a problem with a X100, i do a external call to the asterisk
server . The dialplan its simple answer and hangup..
when it's done , the telephone which i did the call , is in line but
asterisk server is finish.
I'll apreciate all your suggestion. Greetings, txus.
The asterisk output:
-- Executing Hangup("Zap/1-1", "") in new stack
== Spawn
2006 Nov 15
2
Problems with language support
Hi!
I have configured the language support in asterisk to reproduce spanish
prompts. I have lines for it in sip.conf, iax.conf, zapata.conf and
voicemail.conf as shown:
[general]
...
language=es
...
In zaptel.conf
loadzone = es
defaultzone = es
When I check my voicemail I get in the CLI:
-- Playing 'digits/4' (language 'en')
-- Playing 'vm-Old' (language
2010 Aug 27
7
ASterisk CDR file Master.csv
How can we set the CDR Master file to rollover at say 30 Meg and create a new one
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2009 May 28
1
SIP CALL ENCRYPTION
Hello
May i please know if asterisk is now supporting sip call encryption. It
has been a requirement from one of my client to ensure that all
conversation is well secured from any potential sniffers or inside hackers
Please help or suggest any solution that you feel may help
Kind regards
Sam
2009 May 29
2
SIP CALL: RTP ENCRYPTION
> On Thu, May 28, 2009 at 02:00:15PM -0500, research at businesstz.com wrote:
>> Hello
>>
>> May i please know if asterisk is now supporting sip call encryption. It
>> has been a requirement from one of my client to ensure that all
>> conversation is well secured from any potential sniffers or inside
>> hackers
>>
>> I have reviewed and shall
2008 Feb 13
3
Analog DID
Does anyone have any suggestions for connecting analog DID trunks? I have
some small locations that will have 2 analog DID trunks each, the only
solution that I can see will work will be using a channel bank and T1 card,
but it will be close to $1500 to terminate these DID trunks. Was hoping
someone had some experience using an ATA or TDM card and analog DID trunks.
Rhino Channel Bank - $750
4
2007 Jul 16
3
Zaptel 1.2.19 and 1.4.4 released
The Asterisk.org development team has announced the release of Zaptel
versions 1.2.18 and 1.4.4. These releases are maintenance releases that
fix various known issues. See the ChangeLog included in the releases
for a full list of changes. The ChangeLogs are also available
separately on the ftp site.
Both releases are available as a tarball as well as a patch against the
previous release. They
2007 Jul 16
3
Zaptel 1.2.19 and 1.4.4 released
The Asterisk.org development team has announced the release of Zaptel
versions 1.2.18 and 1.4.4. These releases are maintenance releases that
fix various known issues. See the ChangeLog included in the releases
for a full list of changes. The ChangeLogs are also available
separately on the ftp site.
Both releases are available as a tarball as well as a patch against the
previous release. They
2011 Jun 16
1
Web based call back
Hi,
I am looking for a simple solution to do this.
I wish to have the user to enter their preferred method of connection i.e.
for the cheapest solution to their desktop phone or mobile phone, then plan
callfile based on the number that user provided and dial to the user.
Any suggestions?
CK
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2007 Jan 10
2
Random dropped calls...
I have a customer running Asterisk 1.2.13 with Zaptel 1.2.11 that is
having calls dropped. Sometimes you can stay on the phone for a long
time and sometimes the call is dropped within a minute.
There are 9 lines connected to 3 TDM04B cards. The Panasonic Pbx we
replaced did not have this problem. There are 8 SIP phones and 16
analog phones connected to two Astribank-8 units and everyone