the calls disappear from Asterisk (and the people on the calls won't
know the difference). Otherwise, the calls will continue to be bridged
by Asterisk.
Jared,
Is there a debug mode where I can find these specific messages?
Thanks,
Ron
--
Ron Joffe
Siena Tech, Inc.
3319 Willow Glen Drive
We have one connected.
What's your question ?
On Monday 01 December 2008 13:49, Mark Bergen wrote:
Anyone familiar with getting Asterisk 1.4 and Mitel 3300 to play nice
together?
Mark Bergen
Information Systems Manager
Number TEN Architectural Group
Winnipeg - 204.942.0981
Victoria -
Here is my hardware configuration
TELCO --- PRI1 --- PBX --- PRI2 --- Asterisk
The PBX is a Siemens Hicom 200 EX (Model 80)
We are connecting between the PBX and Asterisk using QSIG switch type.
What I want to do is the following:
1. Call comes from TELCO via PRI1 and enters PBX
2. PBX Routes
request to the PBX so Asterisk is
out of the loop?
Thanks,
Ron
--
Ron Joffe
Siena Tech, Inc.
3319 Willow Glen Drive
Oak Hill, VA 20171
(919) 928-0404
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in the past. In this case the number of
PRI's entering the PBX far outweigh the number of PRI's in the Asterisk
server, so it is not an option. I tried to simplify the example.
Any other suggestions ?
Ron
--
Ron Joffe
Siena Tech, Inc.
3319 Willow Glen Drive
Oak Hill, VA 20171
(919) 928-0404
On Saturday 16 August 2008 14:37, Jay R. Ashworth wrote:
TBCT is a feature of LEC/IXC edge switches; there isn't much use for it
in any other context. I don't care if you're using Asterisk to be an
edge switch, but it's a *carrier* feature, by and large.
Certainly in the specific instance
Vitelity provides me with this functionality.
http://www.vitelity.com
Ron
On Thursday 26 June 2008 17:36, Steve Finkelstein wrote:
Hi all,
I was curious if anyone can recommend a company that would work with
small businesses, and capable of using a fallback number (mobile
phone, home
try pri_cpe instead of pri-cpe
On Thursday 19 June 2008 12:51, Eve-Ellen Cole wrote:
I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1
crossover, and I'm currently stuck. Anyone have any thoughts on what I
can do to get past this?
Asterisk side
Digium TE220B w/
On Tuesday 18 March 2008 22:12, Steve Totaro wrote:
For your use, I would go for a RAID 5
I would highly recommend against a raid 5 set. I can give you more details if
you are interested, but these guys have most if it down : www.baarf.com see
the link on the left on why should I not use Raid
On Wednesday 19 March 2008 10:36, Steve Totaro wrote:
And I can post a link that shows a bunch of guys think the earth is
flat with a 5/10 google ranking also (like the barf guys).
http://www.alaska.net/~clund/e_djublonskopf/Flatearthsociety.htm
Steve,
My purpose was to try to point out that
On Sunday 16 March 2008 08:09, broadband Voice wrote:
Cavalier did not leave me any paperwork to sign. I check with the colo as
well and they did not receive anything. I have a feeling I have not defined
my trunk groups well in zaptel.conf.
Your zapata.conf looks ok.
Try this for your
On Sunday 16 March 2008 18:19, broadband Voice wrote:
I tried that and got 14 errors, see below:
Sorry, switch those around, I gave you a zapata.conf, your original
zaptel.conf looks fine.
Ron
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On Saturday 15 March 2008 21:22, Darren Wright wrote:
That's not going to tell you anything about the digits in transit. That's
just telling you that your PRI is up.
you are going to need exten = 4DIGITS
How about trying this out:
exten = _X.,1,NoOp
exten = _X.,n,SOMETHING
exten =
On Tuesday 11 March 2008 16:21, [EMAIL PROTECTED]
wrote:
What is the best alternative for getting the IVR and other prompts recorded
for Asterisk.
We decided to record our own. We set up a recording studio, and that has
worked out very well for us.
Let me know if we can help.
Ron
...can you expand on that please ? I'm on my way to getting one of the
newer Digium TE220B PCIe dual T1/E1 to put on such a system.
I know the subject line was anti-Dell, but just to put in a data point:
We have 10 Dell PE2950's running with one or two TE220B's per system, and they
have
On Friday 01 February 2008 15:31, Matt wrote:
It's about time Digium got on the ball and made PCI-e cards. What are
people's experiences with this card? Anyone know if there are plans for a
PCI-e analog card for FXO use?
I have been using 220B's for about 6 months. I have about 20 of them
On Monday 28 January 2008 14:10, Steve Totaro wrote:
On Jan 28, 2008 1:33 PM, Ron Joffe [EMAIL PROTECTED] wrote:
I have a system with 2 TE220B (2xPRI). I am looking for a method to
shutdown one of the 4 zap lines.
Something like a ifconfig eth3 down command.
It should
I have a system with 2 TE220B (2xPRI). I am looking for a method to shutdown
one of the 4 zap lines.
Something like a ifconfig eth3 down command.
It should be the equivalent of physically unplugging one PRI circuit.
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I have a system with 2 TE220B (2xPRI). I am looking for a method to shutdown
one of the 4 zap lines.
Something like a ifconfig eth3 down command.
It should be the equivalent of physically unplugging one PRI circuit.
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On Tuesday 15 January 2008 12:32, Naveen Palani wrote:
Eg. swift -o /tmp/test.wav -p audio/channels=1,audio/sampling-rate=8000
Press 1 to confirm. Press 3 to cancel.
Naveen,
How about generating the wav files and storing them, then playing the wav's
from the call tree, rather then
On Friday 14 December 2007 14:43, Vincent wrote:
OTOH, having to run a separate PC just to handle calls from a single
POST line AND having to install Linux + Asterisk on this thing... It'd
have to be an appliance (which I haven't seen avaiable in this price
range).
Didn't you just define an
On Saturday 08 September 2007 04:52, satish patel wrote:
I have asterisk server with 2 E1 port now i want to
redendecy for my server means one of server goes down automatically second
goes in active mode is it possible and how to switch E1 to second server ??
On Wednesday 29 August 2007 08:12, equis software wrote:
Hi!
I need to use text to speech, what is the best application?
Thanks!
Cepstral
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To
On Saturday 25 August 2007 11:15, Steve Totaro wrote:
Probably not much help, but if you rarely issue commands such as this,
hit the up arrow a few times.
Steve,
The issue is that I am attempting to restart asterisk from external scripts
due to certain pbx conditions.
I need to have a two
On Saturday 25 August 2007 21:03, Philipp Kempgen wrote:
Maybe something like this:
asterisk -rx 'restart gracefully'
Then constantly monitor
asterisk -rx 'core show uptime seconds'
And if that does not drop to let's say less than 30 seconds:
asterisk -rx 'stop now'
sleep 1
killall -9
On Friday 24 August 2007 12:37, Ryan M. Colbert wrote:
I'd be interesting in pooling resources for this. We've seen the success of
Vonage's Visual Voicemail and would like to emulate a similar solution.
Please define success,
I have a vonage account, and the transcription is very poor at
If I issue a restart gracefully command, the system will wait until all
channels are idle before restarting.
During the time the system is waiting for idle activity, is there a command
that can let me know it is in graceful restart wait mode ?
Thanks,
Ron
On Thursday 23 August 2007 12:46, [EMAIL PROTECTED] wrote:
Thanks for your reply. I have previously looked at the logger.conf file.
I see that the various types of information can be logged in different
ways. After setting the various information types with whatever I want
logged, is it
On Thursday 23 August 2007 14:00, Tzafrir Cohen wrote:
I utilize this command:
nohup script -f -c asterisk -vvvTn /tmp/asterisk.log
To start up my apps. This will log everything to a log file.
Why nohup? And if you have nohup, why script?
It will log everything until the
On Wednesday 22 March 2006 00:33, Nathan Alberti wrote:
Here is a dump of the configuration options, you will see there is a
few new, these are also documented on the wiki.
Nathan,
How did you go about obtaining the dump ?
Thanks,
Ron
___
On Wednesday 22 March 2006 10:01, Nathan Alberti wrote:
Telnet to the phone, login and type show config
Thanks Greg and Nathan!!
Ron
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Anyone have experience with the 3-08-2 release of Cisco's SIP firmware?
Are there any new features in the SIPDefault.cnf?
Thanks,
Ron
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Hey folks,
I have cisco 7940 with SIP configured on Asterisk stable. I can place calls to
it from either my Snom Sip phone on analog extensions on a zaptel card. But I
don't seem to be able to place calls from it to either the snom, or the Zap
extensions.
My log shows:
NOTICE[8923]:
Hey folks
I have a new setup with a TDM400P for a pair of analog extensions and a few
SIP phones. We seem to be experiencing a bunch of Crackeling when talking
between the analog and SIP extensions.
Any ideas?
Thanks,
Ron
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On Monday 14 March 2005 16:18, Eric Wieling wrote:
Ron Joffe wrote:
Hey folks
I have a new setup with a TDM400P for a pair of analog extensions and a
few SIP phones. We seem to be experiencing a bunch of Crackeling when
talking between the analog and SIP extensions.
Any ideas?
Yes
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