Hi,
I am facing problem in configuring D-channel for TE120P card.I did the
following things
/etc/zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
Hi,
Do the following steps are required while configuring D-channel for
TE120P card ( TE120P card is connected through cable to E1 card running
application). These steps are written in TE120P card documnetation.
Hi,
Do the following steps are required while configuring D-channel for
TE120P card. Te120P card is connected to E1 card running application
through cable.The following steps are written in the documentation of
TE120P card
On Sat, Jul 28, 2007 at 12:03:33PM +0530, [EMAIL PROTECTED] wrote:
Hi,
I am facing problem in configuring D-channel for TE120P card.I did the
following things
/etc/zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
Dear Jared;
Thanks a lot for your kindly answer.
Yes, but what does it mean:
Phone/phone0 and Consol/dsp?
Regards
Bilal
On Fri, 2007-07-27 at 06:46 -0700, bilal ghayyad
wrote:
What the following mean:
CONSOLE=Phone/phone0
CONSOLE=Console/dsp
TRUNK=Zap/g2
These are global variables as
On 7/27/07, dave cantera [EMAIL PROTECTED] wrote:
randulo,
I could not get into the conference today... the SIP line was busy, no
matter what I do, the website thinks I'm not logged in and gives me the
login page. after I login, anything I want to do brings me back to the
login page... so I
Sorry if this appears twice - I originally sent it nearly 18 hours ago
and never saw it ..
I have a need to have a unique integer number that can be used by a
dynamic meetme room (I am wanting to redirect a call into a meeting
room, and need a unique number to make sure I don't put two people
Jay Moore wrote (received 2007-07-28):
My boss would like some statistics on how many calls are answered out of
specific queues during a given time period, and I'm not sure how exactly
to obtain those stats. Here's how our queue system works.
1) Call comes in and enters our 'ring' queue
Julian Lyndon-Smith wrote:
Sorry if this appears twice - I originally sent it nearly 18 hours ago
and never saw it ..
I have a need to have a unique integer number that can be used by a
dynamic meetme room (I am wanting to redirect a call into a meeting
room, and need a unique number to
The contents of this e-mail are intended for the named addressee only. It
contains information that may be confidential. Unless you are the named
addressee or an authorized designee, you may not copy or use it, or disclose it
to anyone else. If you received it in error please notify us
Thursday, August 9, 2007, 7:00 PM
http://asteriskpbx.meetup.com/2/calendar/6012673/
See you there
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
Watkins, Bradley wrote:
The contents of this e-mail are intended for the named addressee only. It
contains information that may be confidential. Unless you are the named
addressee or an authorized designee, you may not copy or use it, or disclose
it to anyone else. If you received it in
Nope, that's good. It means you have registered to their server no problem.
Firstly, which version of Asterisk are you using?
Version 1.2.7
If you turn on iax2 debug, and then say call from your cellphone to the
DDI you have registered do you get anything at all?
No, I do not see
Hello Jay,
you may want to have a look at QueueMetrics - everything you're looking
for is already there. :-)
l.
On Thu, 26 Jul 2007 16:37:56 +0200, Jay Moore [EMAIL PROTECTED]
wrote:
Greetings, list!
My boss would like some statistics on how many calls are answered out of
specific
On Fri, 27 Jul 2007, Bertram Scharpf wrote:
Hi,
the first thing I did with Asterisk is listening to
`demo-congrats' by Xlite on the same machine. This works
perfectly. The config files are those shipped with the
package.
Now I want to listen to it over ISDN/Capi but I don't
succeed.
Not sure what you are doing with meetme but,
i Always used AstDB() for this type of needs.
On 7/28/07, Lee Jenkins [EMAIL PROTECTED] wrote:
Watkins, Bradley wrote:
The contents of this e-mail are intended for the named addressee only.
It contains information that may be confidential. Unless
Deepak Naidu wrote:
Hi,
I have a Dell Power Edge server planning yo buy Sangoma A101D
card. To configure with my Asterisk 1.2.18 zaptel-1.2.17.1 Free-PBX
setup.
It would help to know exactly what Dell Poweredge you were considering.
They do vary.
If you compile your kernel
Corporate IT Solutions - Michael Dunne wrote:
I have now within 18 months had a second TDM400P die, the first time was
random call drops, and now it will not go off hook when making a call.
To summarise, the card stopped making calls, I replaced the computer
hardware, installed new OS and new
[EMAIL PROTECTED] wrote:
Thanks for the reply. Unfortunately that didn't work. What's confusing
is that for the line without any distinctive ring that works correctly
with callerid, the only thing it does is dial the phones, so here's the
entire context:
[add-incoming]
exten =
Erick Perez wrote:
Hi,
after many issues we finally managed to make our system do outgoing
calls with perfect quality.
However I cannot detect *any* form of incoming call. when I use an
outside phone to call the E1 connected to the sangoma a102, I
instantly get a fast busy tone.
Let's see
Erick Perez wrote:
Yes I do. I even did a pri debug span 1 and when I call the asterisk
box, it sees nothing.
Hmn, well, that's telling.
Are you using the correct cable? Is the cable plugged into the correct
port on the card? The 102 is a two-port.
-Stephen-
Cheikhou DIAW wrote:
hi , i think everybody is receiving theses mails from rp.
can someone unsubscribe or do something , its really annoying
Now I know why I had 600 messages in my Asterisk folder after only three
days away.
-Stephen-
___
If you do not have any alarms and PRI debug span 1 still gives you
nothing then you need to call your telco and say I'm not getting any
Q.931 messages on the D-Channel.
Stephen Bosch wrote:
Erick Perez wrote:
Yes I do. I even did a pri debug span 1 and when I call the asterisk
box, it sees
Patrick Buller wrote:
Nope, that's good. It means you have registered to their server no problem.
Firstly, which version of Asterisk are you using?
Version 1.2.7
That is super old. Did you install it from a package? I recommend you
upgrade now, because you will have to later, I
On 7/26/07, Patrick Buller [EMAIL PROTECTED] wrote:
This is what callwithus is supposed to forward the call to:
IAX/iaxin:[EMAIL PROTECTED]/[EMAIL PROTECTED]
Does that need to be IAX2/iaxin:[EMAIL PROTECTED]/[EMAIL PROTECTED] ?
Notice the 2? It used to be that IAX referred to v1 of the IAX
Hello,
Do you have porf forwardin for SIP protocol in your firewall?
SIP: 5060 udp
rtp 1 - 2 udp (default)
and IAX2 4569 udp
Best Regards
Carlos Rojas
On 7/28/07, Ary Junior [EMAIL PROTECTED] wrote:
Hi, Im a asterisk newbie and I've configured an asterisk server here in my
aryjunior,
is your dialplan and registration configured to connect to another *
server?...include your config so we can analyze it...
daveC
Carlos Rojas wrote:
Hello,
Do you have porf forwardin for SIP protocol in your firewall?
SIP: 5060 udp
rtp 1 - 2 udp (default)
and
michael,
this is what I use for centOS 4, but I think its too loose... let me
know if you don't know where to put it...
daveC
# for asterisk
-A RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp -m udp --dport 4569 -j ACCEPT IAX
-A RH-Firewall-1-INPUT -p
I am submitting a patch to the Bug tracker next week that will have a manager
event fired alongside every queue log write. You can then send the queue
information to the database in realtime if you have a manager interface script.
If anyone is willing to test this patch once posted, I would
I've been trying to get Asterisk 1.4.X running under Solaris 10 x86 with
limited success.
I can build Asterisk and get it started but have run in to a problem
with a segmentation fault with the help command in the CLI.
When I start Asterisk:
# ./asterisk -vvvgc
Asterisk 1.4.9, Copyright (C)
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