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 get this working I could
see this being my favorite phone
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 get
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,
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
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
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.
___
--Bandwidth and Colocation
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
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
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
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
[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'm
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 where
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'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?
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/ring
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
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
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 providerrefused to doanything (despite the premium SLA) on the basis that
fires weren't covered. I switched
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
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 createthat I can get email(I'm using Postfix mail server)and send to fax machine. Dilemma is that I'm newbie
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 an
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 refused to
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
To UNSUBSCRIBE or update options
Norbert Zawodsky wrote:
quote
... (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.) ...
/quote
O.k. - This answers my first question (if there is disadvantage if we
use only 1 digit
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,
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
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
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. (which
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 CA. Any
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,
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
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 really not
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
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
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
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
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
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
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
-- 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, I've
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