Hi,
I'm using Asterisk 1.2.18 on a Debian Etch box. I've noticed a very
strange fact which causes a bad prob. When I get an inbound call, I make
4 phones ring at the same time, one is a Snom while others are Gigaset
C450IP with _latest firmware_.
When I get a call and answer with the Gigaset, a
Hi,
I had a similar problem when I had signalling set at FXS_KS for my 4 FXS
port TDM400P card. I've read long time ago that the signalling in
Australia (where I am) is FXS_LS, so that solved that for me. Try
different signalling methods, hopefully that will solve your problem.
Apparently,
hello there...
our telecom sold us VoIP-numbering, managed by Nortel MCS
I successfully registered Ekiga to it (
http://sol.chel.skbkontur.ru/ekiga.png)
What exactly do I have to write in sip.conf to make Asterisk register on
this SIP ?
Cheers,
Kate
ooh323c requires You to put the following lines:
disallow=all
allow=ulaw
allow=g723.1
..
otherwise it doesn't work
On 2/24/07, Guillermo Salas M. [EMAIL PROTECTED] wrote:
Solved... installed chan_oh323 :)
Looking for a full time Asterisk tech to work in Melbourne, Australia.
Full time, immediate start.
Anyone?
PaulH
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asterisk-users mailing list
To UNSUBSCRIBE or update
Hi all,
I am a bit out with the Asterisk 1.4.4, after I complied and installed the
Asterisk and I got such error messages
[Jun 28 16:56:19] WARNING[28625] res_smdi.c: No SMDI interfaces are
available to listen on, not starting SDMI listener.
[Jun 28 16:56:19] WARNING[28625] loader.c: Error
hi,
I am installing the asterisk-1.2.19 with zaptel-1.2.8 on FC5.
I got install installed ok.. after i had disable the xpp_usb module.
However, when i try to compile asterisk and having this error
I will be glad for your kind response.
Goksie
chan_zap.c: In function âpri_dchannelâ:
vm_general.conf is where I've set mine (freepbx installation)
D.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Pfeifer
Sent: Thursday, June 28, 2007 12:03 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] voicemail.conf
Hello,
I'm using digium E1 cards and serving SIP users at Asterisk. After the
following call (see below) CDR shows two records. First looks as
outbound call, but the second - as inbound call. Is it a bug or intended
behavior?
Call flow:
SIP (ext: 100) - ZAP (national number)
SIP (ext: 100)
Thanks for your answer Jared, but I also tried that with no luck:
Connected to Asterisk 1.4.5 currently running on moe (pid = 22879)
-- Remote UNIX connection
Verbosity is at least 6
moe*CLI keys show
No such command 'keys show' (type 'help' for help)
Any clue of what can be wrong with my
Hello everybody,
Just was wondering if somebody can help for G711 fax passthrough w/ asterisk.
The issue I have is regarding codec upspeed when the call is already
connected using G729 for example. The setup is
fax---ATA---asterisk---Cisco---fax
When codec upspeed should happen,
Sip registration started working again at 21.46 EDT 6/27/07 so it must
have been a problem with Voicestick - no information on their website
however.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Huw
Richards
Sent: Wednesday, June 27, 2007 20:14
Eeeeck! Asterisk is inserting random digits in dialed numbers.
So far I've seen it insert a 2 after the STD (area) code and insert an
extra 6 or 7 in the STD code. It's pretty repeatable although the
inserted number changes.
My Config is: Asterisk 1.4.5, Zaptel 1.4.3, Digium TE205P (rev 02).
hello there...
our telecom sold us VoIP-numbering, managed by Nortel MCS
I successfully registered Ekiga to it (
http://sol.chel.skbkontur.ru/ekiga.png)
What exactly do I have to write in sip.conf to make Asterisk register on
this SIP ?
Cheers,
Kate
On Thu, Jun 28, 2007 at 11:30:04AM +0100, Goke Aruna wrote:
hi,
I am installing the asterisk-1.2.19 with zaptel-1.2.8 on FC5.
Zaptel 1.2.18, right?
I got install installed ok.. after i had disable the xpp_usb module.
Are you using the default kernel of FC5 (2.6.15) or the one in the
Tzafrir Cohen wrote:
On Thu, Jun 28, 2007 at 11:30:04AM +0100, Goke Aruna wrote:
hi,
I am installing the asterisk-1.2.19 with zaptel-1.2.8 on FC5.
Zaptel 1.2.18, right?
YES .. MY zaptel is 1.2.18
I got install installed ok.. after i had disable the xpp_usb module.
Are you using the
2007/6/27, Greg Oliver [EMAIL PROTECTED]:
On Tue, 2007-06-26 at 21:45 +0200, Olivier wrote:
Hi,
Has anyone met any success, installing localized (ie non-english)
menus within SIP firmware enabled Cisco 7941 ?
Those phones seem to be trying to download localized menus from Cisco
Call
On Thu, Jun 28, 2007 at 01:28:28PM +0100, Goke Aruna wrote:
Tzafrir Cohen wrote:
On Thu, Jun 28, 2007 at 11:30:04AM +0100, Goke Aruna wrote:
hi,
I am installing the asterisk-1.2.19 with zaptel-1.2.8 on FC5.
Zaptel 1.2.18, right?
YES .. MY zaptel is 1.2.18
I got install
Hello All,
Please anyone can help me with this error...
Found some strange problem while Asterisk trying to call the AGI file.
If I pick up the call on the first attempt, it will execute my AGI file
properly.
But if I don't pick up the call and let Asterisk call me again, it adds
StartRetry:
hello,
We looking for a channel bank to connect 120 analogs phones, in SIP to
an Asterisk ..
Did someone have some product in mind.
Thanks for you help..
bye bye
___
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On Jun 28, 2007, at 8:00 AM, pixiesfr wrote:
hello,
We looking for a channel bank to connect 120 analogs phones, in SIP to
an Asterisk ..
Did someone have some product in mind.
A channel bank must connect via a T1 by definition, which would then
give you 24 phone lines per T1. This
We're really happy with SIP Tapi:
http://sourceforge.net/projects/siptapi/
http://www.enum.at/index.php?id=479
We've been trying to document our setup at:
http://projects.bebr.ufl.edu/wiki/AsteriskTAPI
We like it as it requires no manager interface (it uses SIP Refer) to be
turned on. It can be
I did a lot of googling until I found this thread:
http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html
--
Grigoriy Puzankin
Grigoriy Puzankin wrote:
Hello,
I'm using digium E1 cards and serving SIP users at Asterisk. After the
following call (see below) CDR shows two
Hi,
I am trying to establish call through sip phone between two PC connected
to linux box on which asterisk server is running
1st PC is having IP Adress : 192.168.1.149
2nd PC is having IP Adress : 192.168.1.53
Now, I am tying to dial from 1st PC to 2nd PC
Anyone running caller id spoofing applications in the USA running
asterisk?
Then it's time to move them to Canada or similar.
http://arstechnica.com/news.ars/post/20070627-caller-id-spoofing-about-t
o-be-outlawed.html
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello List,
since some days i run into the problem that one span on a TE407P is not
comming up correctly. With intense debug on that span i get:
[ 02 01 7f ]
Unnumbered frame:
SAPI: 00 C/R: 1 EA: 0
TEI: 000EA: 1
M3: 3 P/F: 1 M2:
On Thu, 2007-06-28 at 14:52 +0200, Olivier wrote:
2007/6/27, Greg Oliver [EMAIL PROTECTED]:
On Tue, 2007-06-26 at 21:45 +0200, Olivier wrote:
Hi,
Has anyone met any success, installing localized (ie
non-english)
menus within SIP firmware
On 6/28/07, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi,
I am trying to establish call through sip phone between two
PC connected to linux box on which asterisk server is running
1st PC is having IP Adress : 192.168.1.149
2nd PC is having IP Adress : 192.168.1.53
Remove any relaxdtmf= options from your zapata.conf.
Russell Brown wrote:
Eeeeck! Asterisk is inserting random digits in dialed numbers.
So far I've seen it insert a 2 after the STD (area) code and insert an
extra 6 or 7 in the STD code. It's pretty repeatable although the
inserted
GNUbie wrote:
Hello all,
Anybody can point me to the right URL where I can read an updated manual
for
Asterisk 1.4.x?
Your best bet is to read UPGRADE.txt in the Asterisk source tree. It
should list most of the changes from 1.2.x to 1.4.x
___
El Thu, Jun 28 de 2007 a las 20:18 +0530, [EMAIL PROTECTED] comentaba:
Hi,
I am trying to establish call through sip phone between two PC
connected to linux box on which asterisk server is running
1st PC is having IP Adress : 192.168.1.149
2nd PC is having IP
Hello All,
I have Asterisk 1.4.5 running on a SuSE 10.3 x86_64 2.6.18.2-34
I upgraded from 1.4.2 to 1.4.5 on sunday the 24 of june and since have been
getting:
Internal RTCP NTP clock skew detected: lsr=1402479300, now=1402675136,
dlsr=196500 (2:998ms), diff=664
I see an entry in Mantis that
On Thu, 28 Jun 2007, Jerry Jones wrote:
Your other option would be to do GR303 which would allow you to hang
many lines off a few T1/E1 circuits, except it is definately not SIP.
Well, and it's probably worth pointing out that if you wanted to go the
GR.303 route, the devices on both ends
What is the output of `cat /proc/zaptel/spannumberthatsbroken`
---
Matthew Fredrickson
Software Engineer
Digium, Inc.
On Jun 28, 2007, at 9:53 AM, Alexander Zielke wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello List,
since some days i run into the problem that one span on a
Hello Eric,
On 6/28/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Your best bet is to read UPGRADE.txt in the Asterisk source tree. It
should list most of the changes from 1.2.x to 1.4.x
What I just did was install the asterisk-doc package and I now have the
Asterisk Documentation
You could have the manager interface intiate the call to a local
channel that uses auto answer for your phone. That way it will be
answered automaticaly.
On 6/28/07, Martin Smith [EMAIL PROTECTED] wrote:
We're really happy with SIP Tapi:
http://sourceforge.net/projects/siptapi/
Hi List;
If I need to do a trunk between Asterisk and another
SIP softswitch (so Asterisk will send a SIP calls to
that softswitch), then I have to configure this on the
sip.conf file or where exactly? And is it the same
when I configure iax trunk?
Should I determine the context in this case for
satish patel wrote:
Dear ALL
I want to transfer call from one phone 2 another
phone so this is asterisk feature or SIP Phone feature or endpoint
feature how can i transfer phone call from to another phone
Rgd
Satish patel
Check out this page:
GNUbie wrote:
Hello Eric,
On 6/28/07, *Eric ManxPower Wieling* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Your best bet is to read UPGRADE.txt in the Asterisk source tree. It
should list most of the changes from 1.2.x to 1.4.x
What I just did was install the asterisk-doc
Dean Collins wrote:
Anyone running caller id spoofing applications in the USA running
asterisk?
Then it’s time to move them to Canada or similar.
http://arstechnica.com/news.ars/post/20070627-caller-id-spoofing-about-to-be-outlawed.html
Why it means nothing...
You're a carrier doing
I currently have about 50 polycom 501 phones on my asterisk setup. The
dialplan is set to work with mysql (realtime), and all of the extensions
for the phones route through the same macro (stdexten). This all works
fine until I tried to set up notify status.
On voip-info, they say do something
On 6/28/07, John Millican [EMAIL PROTECTED] wrote:
Would i be correct in assuming that if i pull a copy of
1.4.5 from digium this weekend that this message will go away?
No... you'd have to pull the latest code from the 1.4 branch using
Subversion, or wait for 1.4.6 to be released.
-Jared
It was due to changes in cdr in asterisk 1.4.5 previous version does not do
it .there is a fix on bugs.digium.com or you can wait till next release or
use asterisk 1.4.4
On 28/06/07, Rob Schall [EMAIL PROTECTED] wrote:
I currently have about 50 polycom 501 phones on my asterisk setup. The
Hi
iam using 1.2.X SVN
iam keep getting the below message
Jun 28 23:07:31 WARNING[2692]: channel.c:785 channel_find_locked: Avoided
deadlock for '0x864e70', 10 retries!
any help
ram
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--Bandwidth and Colocation Provided by
What was due to changes? I didn't read anything in the release notes
about hinting in any newer versions (changes, etc). Do you have a link
to this fix? and will this fix work with 1.2?
Rob
Jaswinder Singh wrote:
It was due to changes in cdr in asterisk 1.4.5 previous version does
not do it
On Thu, Jun 28, 2007 at 10:59:34AM +1000, Nathan Dennis wrote:
Thanks Tzafrir, that did the trick.
But please note the that the bristuff patch from xorcom has broken
links in it.
http://updates.xorcom.com/astribak/bristuff ? Updated and fixed, thanks
for the note.
--
I have installed the Asterisk BE B.2.2 image file in a new server. I need to
make network routing changes. However in their version of rPath (pound key)
Digium has removed the netconfig command. I am able to manually add the route
with
Route add default gw xxx.xxx.xxx.xxx however when I
Hi
I have a client using a Avaya IP Office PBX that is taking a SIP trunk
from me terminating on a * box. It all works perfectly apart from DTMF.
Although you can hear the tones they don't seem to get recognised. I
have tried DTMF mode auto, inband, out of band and rfc2833 but no luck.
Any ideas?
In October of last year Marc Blanchet of the Canadian company Viagénie
made a presentation on how he and others had build IPv6 support into
Asterisk and furthermore demonstrated that it worked. Marc Blanchet
went into some details on how it was done and the amount of work that
had gone into it.
A
try to edit /etc/sysconfig/network-scripts/ifcfg-eth0 if u have eth0
if not try ifcfg-eth1 for eth1
On 6/29/07, Ed Nuñez [EMAIL PROTECTED] wrote:
I have installed the Asterisk BE B.2.2 image file in a new server. I
need to make network routing changes. However in their version of rPath
GNUbie wrote:
Hello Eric,
On 6/28/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Your best bet is to read UPGRADE.txt in the Asterisk source tree. It
should list most of the changes from 1.2.x to 1.4.x
What I just did was install the asterisk-doc package and I now have the
Rob Schall wrote:
I currently have about 50 polycom 501 phones on my asterisk setup. The
dialplan is set to work with mysql (realtime), and all of the extensions
for the phones route through the same macro (stdexten). This all works
fine until I tried to set up notify status.
On
If SLA supports IP trunks, can shared extension appearance be achieved using
a local SIP trunk in place of an extension?
Basically, I'm trying to allow some stations (Polycom IP 650) to have a
shared extension amongst all of them. Ideally, I'd like for the LED to show
if that extension is in
Hi,
I would like to set up in the Asterisk system (downloaded from Nerdvittles)
a robo-dialer for an outbound call center. Idea is that the dialer should do
predictive dialing and once the call is answered pass it through to the next
free agent. CTI would be a nice to have. ;-)
Anyone who can
Hi all,
I'm looking for an asterisk user (can be a n00b who knows enough about
asterisk to ask intelligent questions or a brilliant specialist) to
talk about what they do with asterisk. I would like to have a co-host
next week, someone who uses video via the web (it's a Flash
application that can
Hello all,
I'm looking for software for my asterisk logs that will compile the
information into nice web-based charts and graphs. Something that works
similar to webalizer for apache. I want to be able to spot trends of
usage, call volume levels, disconnect/failure levels, and basically
Eric ManxPower Wieling wrote:
Rob Schall wrote:
I currently have about 50 polycom 501 phones on my asterisk setup. The
dialplan is set to work with mysql (realtime), and all of the extensions
for the phones route through the same macro (stdexten). This all works
fine until I tried to set
This allows me to edit the IP Address of the NIC card, but not edit my IP
routing.
Thanks
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ~Russell
Sent: Thursday, June 28, 2007 2:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
I'll start by saying I'm a trixbox user, and a new one at that, so
hopefully you can respond to me on those terms.
I have a user who works from home 1 day a week. On that day I'd like
for him to be able to connect with a softphone and be reachable by just
dialing his extension as we normally
use folow-me
- Original Message -
From: Ryan Stille [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, June 28, 2007 4:16 PM
Subject: [asterisk-users] setup multiple phones for 1 extension
I'll start by saying I'm a trixbox user, and a new one at that, so
hopefully
How many GW you need to add ? if it is one .. then add
GATEWAY=xxx.xxx.xxx.xxx into /etc/sysconfig/network
thanks
Russell
On 6/29/07, Ed Nuñez [EMAIL PROTECTED] wrote:
I have installed the Asterisk BE B.2.2 image file in a new server. I
need to make network routing changes. However in
He can not have the same username/secret. In trixbox - your ring group idea
is probably best...
On 6/28/07, Ryan Stille [EMAIL PROTECTED] wrote:
I'll start by saying I'm a trixbox user, and a new one at that, so
hopefully you can respond to me on those terms.
I have a user who works from
Hi.
Maybe Asterisk Stat could help you.
http://www.areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54
Hope it helps you
Regards
Iván Cetta.
On 6/28/07, Mark Coccimiglio [EMAIL PROTECTED] wrote:
Hello all,
I'm looking for software for my asterisk logs that will compile
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Thursday, June 28, 2007 3:13 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] network routing
This allows me to edit the IP Address of
SB == Stephen Bosch [EMAIL PROTECTED] writes:
SB Hi, folks: I remain intrigued by the gap in BRI implementation
SB between North America and Europe, and I wanted to get feedback
SB from the list members on the matter. I'm seriously considering
SB making the leap in our office.
BRI is being
On Thursday June 28 2007 1:19 pm, Jared Smith wrote:
On 6/28/07, John Millican [EMAIL PROTECTED] wrote:
Would i be correct in assuming that if i pull a copy of
1.4.5 from digium this weekend that this message will go away?
No... you'd have to pull the latest code from the 1.4 branch using
Just add the softphone to the dial command. If it's not connected
nothing will bad happen
and the regular phone will ring. Whenever the softphione is registered
it will ring as well. If the other phone is a SIP phone, you could use
IAX as the softphone with the same username and password.
Thanks, that worked
· I was using GATEWAYDEV=eth1
And that was not working.
Thanks again
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ~Russell
Sent: Thursday, June 28, 2007 3:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
you would think the telcos would be more interested in selling this to
small/medium businesses that are not ready for a voice pri but it
Since when to the telcos have the consumer's best interest in mind? They can
sell you a PRI at full loop cost with a smaller number of channels in the
Greg wrote:
So, if you ever use a Cisco SIP Phone with an Asterisk
server, it's not possible to localize menus, soft
keys, and so on ?
Not unless someone wants to add support for it in the SIP
channel, which I doubt. I would be more than willing to
provide the SIP messages that a
Jeremy Mann wrote:
you would think the telcos would be more interested in selling this
to small/medium businesses that are not ready for a voice pri but
it
Since when to the telcos have the consumer's best interest in mind?
They can sell you a PRI at full loop cost with a smaller number of
I installed the follow-me module and tried it out, it works great. I am
just continually amazed at what asterisk can do.
Another question - I'd like one of the extensions to ring out to a cell
phone. I may have the users press '9' or maybe tell them to use
extension 900 or something, not
Quoting Stephen Bosch [EMAIL PROTECTED]:
Jeremy Mann wrote:
you would think the telcos would be more interested in selling this
to small/medium businesses that are not ready for a voice pri but
it
Since when to the telcos have the consumer's best interest in mind?
They can sell you a PRI
At 02:37 PM 6/28/2007, you wrote:
SB == Stephen Bosch [EMAIL PROTECTED] writes:
SB Hi, folks: I remain intrigued by the gap in BRI implementation
SB between North America and Europe, and I wanted to get feedback
SB from the list members on the matter. I'm seriously considering
SB making the
or in the same file you can just do a
X.X.X.X via Y.Y.Y.Y
Each new one on a seperate line.
On 6/28/07, Watkins, Bradley [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Thursday, June 28, 2007 3:13 PM
On 11/22/06, Xue Liangliang [EMAIL PROTECTED] wrote:
Hi, we are using asterisk 1.2.13. When callbacklogin agent transfer
call using SIP phone's transfer feature, he is always in busy status
and cannot answer any more incoming call from queue until the
transferee hang up the call.
I'm
You can use Queues. Put them in a queue and let them listen to music on
hold.
Cheers,
Dimitri
GNUbie wrote:
Hello all,
I'm running Asterisk 1.4.5 and Zaptel 1.4.3 on Debian Etch i386 with the
Digium's Dev Kit that comes with 1 FXO and 1 FXS. How do I configure my
home PBX in such a way
Nice solution Eric, thanks. Very elegant.
On 6/27/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Matthew Brothers wrote:
Hi,
I have been looking for an example of accomplishing this, but
I've been unable to locate something similar to what I'm trying
to do.
Here's the scenario:
I have been looking into how to setup distinctive ringing on a SPA-2100. So
far the only thing i have been able to find is how to define a distinctive
ring in the spa config. What i cannot figure out is what SIP message i need
to be sending to it in order for it use the ring. I did find out how
Hello Dimitri,
On 6/29/07, Dimitri Volski [EMAIL PROTECTED] wrote:
You can use Queues. Put them in a queue and let them listen to music on
hold.
How do you do this based on my original /etc/asterisk/extensions.conf that I
have on my home PBX? I just want that the PSTN caller will hear a
I did find out how to add the sip message for distinctive ring
i just dont know what variable needs to be passed in
order for it to work.
Try: SetVar(_ALERT_INFO=Bellcore-r2);
etc.
___
--Bandwidth and Colocation Provided by
Hi Bilal -
If I need to do a trunk between Asterisk and another
SIP softswitch (so Asterisk will send a SIP calls to
that softswitch), then I have to configure this on the
sip.conf file
Yes.
And is it the same
when I configure iax trunk?
Not exactly the same, but very close. Here's a
Mike Ryan wrote:
My question is: Can SLA give me the same results? And if so, does it make
more sense to use SLA to achieve this? Lastly, if I use SLA, will I also
have the ability to barge and will I be able to park using the hold button?
The SLA code that is in Asterisk now will not
Bent Bagger wrote:
When will these additions make their way into the Asterisk mainstream
It has not yet been merged into the main development tree, but I'm sure it will
be before Asterisk 1.6 is released.
--
Russell Bryant
Software Engineer
Digium, Inc.
I am not sure what exactly you wish to achieve. Just a basic SIP--to--SIP call
or ?
I am not much into the configs, but ya I can tell you that you can try using
FreePBX or Trixbox kind of setup to write ur Asterisk config file, rather then
u editing them, as it has macros, context etc...
Rob Schall wrote:
Eric ManxPower Wieling wrote:
Rob Schall wrote:
I currently have about 50 polycom 501 phones on my asterisk setup.
The dialplan is set to work with mysql (realtime), and all of the
extensions for the phones route through the same macro (stdexten).
This all works fine
On Jun 22, 2007, at 3:43 PM, Joe acquisto wrote:
I have an existing Hylafax system using a mainpine 4 port board, 4
POTS lines.
Have a recently installed Asterisk system, with a dedicated T1
line. (Well, it's really a fonality system).
What would I need to do, or where is the reading
Sorry i didnt read your mail properly . I thought your problem is with
cdr's. Here's link to cdr problem :)
http://lists.digium.com/pipermail/asterisk-dev/2007-June/028085.html
see the next message for patch .
On 29/06/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Rob Schall wrote:
88 matches
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