On Sat, 2006-09-30 at 00:47 -0600, Colin Anderson wrote:
> yeah, weird that, i did set the proxy after I posted so it's now
> sip:[EMAIL PROTECTED] but still no dice. There is a port number as well
> that I left at 0, should I change it to 5060?
>
> btw, the hardware is a sweet little package if I
On Sat, 2006-09-30 at 09:35 +0200, Dave Cotton wrote:
> and 00085D183552.cfg (not uppercase) contains
Whoops note
--
Dave Cotton <[EMAIL PROTECTED]>
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Hi all,I have EMG 202 dailup VoIP Gateway. Which is not registering to our Asterisk server and generating following debug.Unable to find key '113685' in family 'SIP/Registry'I stored 113685 user and password in MySQL, but first time when i connect to the dialup its registered and once i made first
As in other than asterisk queues telling them automatically? Mine tells
them number of callers and estimated hold time. No third party needed,
standard feature.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rick Smith
Sent: Wednesday, September 27, 200
On 28 Sep 2006, at 19:18, Mark Farver wrote:
Here's my setup, I'd like to hear what everyone suggests is the
best route:
I have three sites, a main and two outlying buildign, each has a
Mitel SX200 system with a PRI interface. The outlying sites
currently each have a PTP T1 line back to
On 27 Sep 2006, at 21:03, [EMAIL PROTECTED] wrote:
1 - Yes we are using the HelloAgiScript from asterisk-java.
Therefore the
fastagi-mapping.properties looks like: hello.agi = HelloAgiScript
2 - The command line, which is build in WinXP
Thx for your help
ps: How can i answer without opening
On 29 Sep 2006, at 05:03, stan ford wrote:
if you have to setup an office of 100 users now. would you rather
setup a sip trunk,a t1-pri, or even a t1? and why?
Both. Get a t1-pri - this will give you high uptime, high quality
calls and all the rest.
Then sign up with a couple of SIP (or
I've forgotten the user/pw for my freepbx adnim. I'm using
[EMAIL PROTECTED] Is there a way to discover them or reset them. I have
root access to the system. I did a google search but that didn't help.
Thanks,
Jim.
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On 29 Sep 2006, at 19:20, Yu Safin wrote:
Hi, I am a salesman currently using asterisk to contact my customers.
So far, I have asterisk connected to two PSTN analog lines where I
only receive phones calls.
Then, I have asterisk connected to a VoIP service company for
terminating my phone calls.
On 28 Sep 2006, at 21:39, Wolfgang_Borgon wrote:
David,
Yes, I've also forwarded port 4569 to the server.
Since the router is forwarding to the server, I cannot
forward it to the client as well -- however, as the
client isn't going out past the LAN, it shouldn't
matter... unless there's somethi
Jim Lynch wrote:
I've forgotten the user/pw for my freepbx adnim. I'm using
[EMAIL PROTECTED] Is there a way to discover them or reset them. I have
root access to the system. I did a google search but that didn't help.
On a normal installation the passwords are located in the voicemail.co
On 29 Sep 2006, at 19:40, Jay R. Ashworth wrote:
On Fri, Sep 29, 2006 at 09:37:06AM +0100, Conrad Wood wrote:
7. Relationship with provider. What is their SLA? Is it the
incumbent or the clec? An incumbent will be more expensive
and more difficult to deal with but they will tend to be more
rel
[EMAIL PROTECTED] wrote:
Hi All,
I'm having trouble detecting faxes reliably. I'm using one analog line for
From what I've read, it isn't reliable.
Doug
-- Ben Franklin quote: "Those who would give up Essential Liberty to
purchase a little Temporary Safety, deserve neither Liberty nor
On Fri, 29 Sep 2006, Lacy Moore - Aspendora wrote:
> I have a suggestion regarding dial plan. When I first started I saw no
> reason to have to dial 9 first for outside calls. Because I wanted to be
> able to dial out from the missed calls list, I chose to eliminate the dial 9
> requirement. I'
Please, I'm having this same problem, but didn't find any solution for this yet.
What version are you using?
How can i disable VAD in a softphone using g729a codec???
Is there a configuration for asterisk.conf or some other file to fix this???
Please, help.
> Noc Phibee wrote:
> >> anyone know wh
On Sat, Sep 30, 2006 at 07:12:23AM -0400, Jim Lynch wrote:
> I've forgotten the user/pw for my freepbx adnim. I'm using
> [EMAIL PROTECTED] Is there a way to discover them or reset them. I have
> root access to the system. I did a google search but that didn't help.
This is not an asterisk i
I'm looking for the username/password to access the web gui for freepbx
admin rather than the voicemail passwords. I need to reconfigure the
extentions/ring groups.
Thanks,
Jim.
Doug Lytle wrote:
Jim Lynch wrote:
I've forgotten the user/pw for my freepbx adnim. I'm using
[EMAIL PROTECTED]
At 12:15 AM 9/28/2006 -0700, you wrote:
i can't for the life of me find a pay as you go termination and
origination service.
there's garfachi, but they don't offer DID's in anywhere else other than
CA. Any suggestions? Thanks.
They do offer tollfree DID. What about using one of those?
Rega
login as root and type help-aah and you will see a list of commands to change the admin password.On 9/30/06, Jim Lynch <
[EMAIL PROTECTED]> wrote:I'm looking for the username/password to access the web gui for freepbx
admin rather than the voicemail passwords. I need to reconfigure theextentions/r
when i make the call ,
on the xlite side i see the call connected but for the sip gateway the call is
ringing and even the phone (PSTN side) is ringing.
I thing that is only
Asterisk send to xlite the signal of connect .
Is there any
configuration to set ??
Thanks
Date: Sat, 30 Sep 2
Has anyone noticed any anamolies with Monitor not recording in 1.4 beta2? I just did a half hour interview this morning, and for the FIRST time ever, Asterisk dropped the recording. The same also happened with a friend yesterday. I don't like this, because I RELY on Asterisk to do my interviews
On 9/30/06, Tim Panton <[EMAIL PROTECTED]> wrote:
Just to amplify this point. I've tried to claim on an SLA. Ourinternet connection was down for a week due to a fire inBT's exchange. My provider refused to doanything (despite the premium SLA) on the basis that
fires weren't covered. I switched pro
Ronald Lewis wrote:
ever, Asterisk dropped the recording. The same also happened with a
friend yesterday. I don't like this, because I RELY on Asterisk to do
Sorry, but I've gotta say it.
Then you shouldn't be using BETA software in production.
Doug
--
Ben Franklin quote:
"Those who woul
Hi,
We are writing an broadcaster software and under mid-heavy load
(dialing on 3 PRIs) we are seeing channels becomming unavailable one
after the other untill no more channels are available to dial on.
Any ideas or hints? We are running latesr libpri, zaptel and asterisk.
Andre
__
Hi,
I'm trying to create fax server with asterisk and spandsp. My situation is: now I can get faxes from fax machines and send tiff files attachments to users email boxes. But now want to create that I can get email (I'm using Postfix mail server) and send to fax machine. Dilemma is that I'm newb
On Sat, Sep 30, 2006 at 01:40:09PM +0100, Gordon Henderson wrote:
> Here in the UK, I've installed several small systems without a dial-9 for
> an outside line type thing. The outside line prefix is effectively digit
> zero. (which is preserved and dialled on the outgoing zap lines)
>
> There is a
On Sat, Sep 30, 2006 at 11:14:08AM -0500, Brandon Galbraith wrote:
>On 9/30/06, Tim Panton <[EMAIL PROTECTED]> wrote:
>
> Just to amplify this point. I've tried to claim on an SLA. Our
> internet connection was down for a week due to a fire in
> BT's exchange. My provider refus
Hello All
Does anybody know where I can find information on
configuring Asterisk 1.4 to work with Google talk.
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asterisk-users mailing list
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Norbert Zawodsky wrote:
... (Well, almost. Extensions must be shorter than 80 characters long,
and you shouldn’t use single-character extensions for your own
use, as they’re reserved.) ...
O.k. - This answers my first question ("if there is disadvantage if we
use only 1 digit extensions").
Bu
Lacy Moore - Aspendora wrote:
I have a suggestion regarding dial plan. When I first started I saw no
reason to have to dial 9 first for outside calls. Because I wanted to
be able to dial out from the missed calls list, I chose to eliminate the
dial 9 requirement. I'm now regretting it, prim
On 30 Sep 2006, at 19:35, Jay R. Ashworth wrote:
I know that this has been a problem for traditional PBXen for years,
and the only solution I've ever been able to see is "use 8 as your
outdial prefix"... but no one seems to ever do that, even 20 years on.
Never say no one. Our legacy PBX is s
I need to build wanpipe to suport a Sangoma a200 card and when I do a
Setup install as directed I get:
Kernel source directory /lib/modules/2.6.9-42.0.2.EL/build not found
I did a yum search kernel but didn't see anything that suggested the
kernel source tree. Running the latest Tribox on cent
On Sat, Sep 30, 2006 at 02:35:49PM -0400, Jay R. Ashworth wrote:
> On Sat, Sep 30, 2006 at 01:40:09PM +0100, Gordon Henderson wrote:
> > Here in the UK, I've installed several small systems without a dial-9 for
> > an outside line type thing. The outside line prefix is effectively digit
> > zero. (
Check it!
http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk
Robert LaPoint escribió:
Hello All
Does anybody know where I can find information on configuring Asterisk
1.4 to work with Google talk.
--
no, i have to retain my fax numbers hmm. thanks anyways.John Kington <[EMAIL PROTECTED]> wrote: At 12:15 AM 9/28/2006 -0700, you wrote:>i can't for the life of me find a pay as you go termination and >origination service.>>there's garfachi, but they don't offer DID's in anywhere else other than >C
I have already tried to follow this document but it did not work under 1.4,
so I am just wondering if Google talk is even supported under asterisk 1.4
yet.
Thanks Alberto
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alberto
Sagredo
Sent: Saturday, Sept
What did not work? I made test under SVN Trunk and only have issues with
audio behind NAT clients.
You could check at bugs.digium.com Gtalk development state and bugs
resolved.
I did not make test with 1.4 beta 2 , so i could not help you more
Regards
Robert LaPoint escribió:
I have alread
On Sat, Sep 30, 2006 at 10:39:09PM +0300, Tzafrir Cohen wrote:
> > I know that this has been a problem for traditional PBXen for years,
> > and the only solution I've ever been able to see is "use 8 as your
> > outdial prefix"... but no one seems to ever do that, even 20 years on.
> >
> > Is this
On 30 Sep 2006, at 20:37, Jim Lynch wrote:
I need to build wanpipe to suport a Sangoma a200 card and when I do
a Setup install as directed I get:
Kernel source directory /lib/modules/2.6.9-42.0.2.EL/build not found
I did a yum search kernel but didn't see anything that suggested
the kernel
With all due respect, if you are 'rely'ing on your phone system for business
you should not be using beta sofware.
-Original Message-
From: Ronald Lewis [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 30, 2006 9:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
The Digium G.729 binary codec modules (both 32-bit and 64-bit), and register
tool have been updated for use with the Asterisk 1.4 beta on Solaris.
As always, they can be downloaded from
http://ftp.digium.com/pub/telephony/asterisk/g729/
--
Jason Parker
Digium
On Thu, 2006-09-28 at 16:54 -0600, Colin Anderson wrote:
> Erm, I think what the OP was referring to was something like this:
>
>
> ____
> _
>
> A. SIP service-->B. His Asterisk install->C. His
> customer's i
oj that's awesome info thanks i will try it when i get back to the office on
monday (hey even geeks have to ride bikes on the weekend esp with the leaves
turning) fwiw i used the web interface and it's as brutal as a Grandstream
but other than that this little number looks *so* sweet - cordless ran
I've used various versions of Asterisk for many things ... this isn't necessarily a "production" thing. I'm fully aware of the nature of beta software (I've tested a lot of software in my time), and I'm simply asking for feedback ... right now, this type of feedback doesn't help, but thanks anyway.
Abdul wrote:
But the same EMG has Broadband inteface once i use this the problem is
never happened and always it registered to asterisk. So the problem is
only for the dialup interface.
I think you should take it up with your supplier, after all they're the
ones taking your money. It is not a
I am having issues with a new installation, configure completes successfully, then once a "make" is run, the following errors show;configure: creating ./config.statusconfig.status: creating build_tools/menuselect-deps
config.status: creating makeoptsconfig.status: creating channels/h323/Makefilecon
Sure sounds like a firewall issue... if you pinging port 4069 and it is not coming back, that sounds like a firewall problem. Try taking down your iptables and then try & see what happens.
bp
On 9/28/06, Wolfgang_Borgon <[EMAIL PROTECTED]> wrote:
David,Yes, I've also forwarded port 4569 to the s
-- Forwarded message --From: "Jay R. Ashworth" <
[EMAIL PROTECTED]>To: asterisk-users@lists.digium.comDate: Sat, 30 Sep 2006 14:35:49 -0400
Subject: [asterisk-users] Dial-9 (was Extension Numbering)
On Sat, Sep 30, 2006 at 01:40:09PM +0100, Gordon Henderson wrote:> Here in the UK,
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