I have a site running 1.4.17 with Zaptel. They want to add a TE420P for
additional T1 capacity. I'm 99% sure this will work, anyone aware of a
reason it wont?
Thanks,
James
--
_
-- Bandwidth and Colocation Provided by
I found that on a clean boot, I could not connect to Postgresql either.
In my /etc/rc.local, I unload cdr_pgsql.so, sleep 15, then reload the
module, and that seems to work. After bootup, cdr_pgsql.so is able to
connect immediately.
--
James Texter III
Sr. Software Engineer
NOBLE SYSTEMS
4151
setup.
--
James Texter III
Sr. Software Engineer
NOBLE SYSTEMS
4151 Ashford Dunwoody Road, Suite 600 | Atlanta, GA 30319-1452
(o) 404.851.1331 ext. 357
(f) 404.851.1421
(e) jtext...@noblesys.com
(w) www.noblesys.com
We succeed when we exceed our customers expectations!
-Original Message
I received this with a Sangoma card and CentOS 5.4. Downgrading to 5.2
resolved the issue.
--
James Texter III
Sr. Software Engineer
NOBLE SYSTEMS
4151 Ashford Dunwoody Road, Suite 600 | Atlanta, GA 30319-1452
(o) 404.851.1331 ext. 357
(f) 404.851.1421
(e) jtext...@noblesys.com
(w
Try putting in a wait after you answer. It's possible the message is
playing before the RTP is setup. I would change your dialplan to be
exten = 333,1,Answer()
exten = 333,n,Wait(1)
exten = 333,n,Playback(vm-goodbye)
exten = 333,n,Hangup()
HTH,
James
On Mar 17, 2008, at 5:47 AM, Anselm
Thunderbird):
http://www.unison.com/opensource/
Thanks,
James Texter
On Mar 11, 2008, at 8:59 AM, Tzafrir Cohen wrote:
On Tue, Mar 11, 2008 at 09:48:04AM -0400, Dean Collins wrote:
http://www.pcworld.com/article/id,143198-pg,1/article.html
anyone know anything about it?
No, but I have
incoming calls are coming top down, then you need to
use Gyour group number in your Dial app so that outbound calls go
bottom up, or vice versa.
HTH,
James Texter
On Feb 27, 2008, at 2:55 PM, Tim Nelson wrote:
Hello! I've run into a problem where a user is making an outbound
call
,
but any remote workers connect via VPN, and the phone registers on the
data network.
Thanks,
James Texter
On Nov 28, 2007, at 3:58 AM, Chris Bagnall wrote:
Greetings list,
I remember a discussion many months ago which ISTR concluded that
asterisk didn't play nicely at all in multi-homed
Hello listers,
I went to pull some CDR's from my PBX, and noticed they were a bit
light. I also noticed output on the console about CDR's not being
posted. I am currently running 1.4.13, and in looking at the change
log, this was a change in behavior as part of mantis 10659. Personally,
I believe libtool-ltdl-devel is what you need.
On Mon, 2007-10-01 at 13:22 -0500, Chris Stinson wrote:
The libtool-ltdl package is installed.
On 10/1/07, Jared Smith [EMAIL PROTECTED] wrote:
On Mon, 2007-10-01 at 12:52 -0500, Chris Stinson wrote:
I'm having an error when I try to
What do you have ulimit -n and ulimit -x set to?
Thanks,
James Texter
On Fri, 2007-09-21 at 08:51 -0400, Wai Wu wrote:
I am not so sure if the interrupts has any thing to do with it. I run some
more test just now and I am getting these error on the console of the call
receiving machine
Have you tried setting resetinterval=never in zapata.conf?
On Tue, 2007-07-17 at 15:43 +0200, [EMAIL PROTECTED] wrote:
Hi,
Lately we've noticed that some Zap channels on one of our PRIs are
unavailable. We have 2 PRI lines with 60 channels in total. On the first
PRI there are currently 20
Do you have the mysql client and header files installed?
On Thu, 2007-06-21 at 04:11 -0700, Khaled Chehab wrote:
Yes mysql installed
[EMAIL PROTECTED] asterisk-1.4.5]# rpm -q mysql
mysql-4.1.20-2.RHEL4.1
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Have you checked to ensure the card in server #2 is jumpered for E1?
On Thu, 2007-06-21 at 09:34 -0500, Jason K. Carter wrote:
Hi there,
I've got two Asterisk hosted PBX servers with Digium TE210P cards
attached by a E1 cable to Port 1 on each. On startup, both cards flash
red,
If you do make config when compiling zaptel and asterisk, it should
put the script in /etc/init.d, and add the relevant entries to the
various start levels.
Thanks,
James Texter
On Fri, 2007-05-04 at 18:44 +0200, Christian wrote:
Hi,
I have already done:
apt-get build-dep asterisk
I sent this yesterday, but saw zero traffic, so I think it got lost in
the ether, so I'm sending again.
I'm having an issue using Asterisk 1.2.14 and an Audiocodes Mediant 1000
ISDN gateway. For the most part, everything is working except for
attended transfers. When I do an attended transfer,
On Mon, 2007-01-15 at 15:26 -0600, David Gomillion wrote:
I don't think you can do that. Here's why: on the Polycom's, the
Transfer button doesn't reappear until the transferree picks up the
phone. Unless something changed in the firmware recently. But, if you're
completing it before the 3rd party
it?
Good grief!
Doug.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
James Texter
list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
James Texter
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit
why it might be happening? Thanks,
Mike
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
James Texter
and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
James Texter
___
--Bandwidth and Colocation provided by Easynews.com
by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
James Texter
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
Wieling [EMAIL PROTECTED] wrote:
Best of luck getting multiple instances of Asterisk to play nice when
accessing Zap channels.
James Texter wrote:
Doug,
I actually see this as a pretty logical way to solve the problem.
Please keep us posted if you have any luck sorting out running multiple
and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
James Texter
___
--Bandwidth and Colocation provided by Easynews.com
.
--
James Texter
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
for stopping on a specific digit, but is there an existing method I can call that will only stop playing if received a specified number of digits?
Thanks,
--
James Texter
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
Title: Re: [asterisk-users] Polycom ACD, Asterisk, Kernel 2.6
What OS are you using? There is a known issue with the kernel sources on CentOS 4.3 and I assume RHEL 4 that will keep Zaptel from compiling? What compilation error are you getting?
James
On 7/11/06 3:26 PM, Dean @ INKnBITs [EMAIL
Title: Re: [Asterisk-Users] when I press transfer - blind - 700 . The user is not able to hear what extension the call was parked on
This is the way blind transfers work. The transferring party doesnt get to hear anything. For call parking, you have no choice but to use supervised transfer if
Title: Re: [Asterisk-Users] Echo Problem with T411P
Try setting echocancelwhenbridged=yes. Also, in your zaptel, you only need to define one span as the clock master, so should be like
zaptel.conf--
span = 1,1,0,ccs,hdb3,crc4
bchan = 1-15,17-31
dchan = 16
span =
Title: Re: [Asterisk-Users] Problems with zaptel and TE210P
Shouldnt your zapata.conf be
span=1,1,0,esf,b8zs
As it stands, you are not taking timing from the PRI. Changing the second digit of the span entry to 1 will tell Asterisk to use that line as the clock master.
HTH,
James
On
Has anyone been able to overcome the limit of being able to watch 7 people?
If so, how?
On 4/3/06 5:52 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Mon, 3 Apr 2006, Darrick Hartman wrote:
Very good explanation. Additionally, (at least on the Polycom 600's) you
need to reboot your phone
I have a 4 PCI slot version. It worked well under Windows, but I could
never get my laptop to see any devices I stuck in it when running under
Linux. I have tried RHEL3 and RHEL4. They list their last supported
Linux version has Redhat 9, so doesn't appear they keep the Linux
version very up
that change was made, I have not had any
other issues.
Thanks,
James
James Texter wrote:
So, I'm still having this problem with outbound calls not working when using a channel
bank. I've purchased a Rhino FXO channel bank from VoIPSupply.com to make sure it wasn't
an equipment problem. I am using
I'm hooked up to a regular analog POTS line. I've tried both loop start
and ground start, but no luck either way. Any other thoughts?
Thanks,
James
Doug Lytle wrote:
[EMAIL PROTECTED] wrote:
So, I'm still having this problem with outbound calls not working
when using a channel bank.
In my telecom experience, overlap on a PRI isn't sending digits as
INFORMATION messages, but instead means to send the digits as DTMF tones
over the B channel. This is pretty common when connecting to the PSTN
where the carrier requires an authorization code, either for billing or
as an access
35 matches
Mail list logo