e messages, and they seem
to be comming from only certain type of ATA's.
I'm suspecting it's ATA related, but I don't have enough evidence to
prove so yet.
Andy
On 3/14/06, Dan Morin <[EMAIL PROTECTED]> wrote:
>
>
>
> The past two days, I've been havi
The past two days,
I've been having issues with my two VoIP service providers where calls just
suddenly hang up. The following is from the log:
Mar 14 13:50:55 WARNING[5887] chan_iax2.c: Max retries exceeded to host
64.34.45.100 on IAX2/voipjet-3 (type = 6, subclass = 11, ts=25,
seqn
Sorry to bring up this old topic, but I had the same issue. The
solution, at least to my problem, was the realization the Teliax lets
you set the codec settings for SIP and IAX independently and the default
setting when you load the page is SIP. So if you make the changes
there, but you're using
@lists.digium.com
Subject: [Asterisk-Users] Re: Horrible MeetMe performance
In article
<[EMAIL PROTECTED]>,
Dan Morin <[EMAIL PROTECTED]> wrote:
> Make sure that if you're using anything other than zaptel hardware, it
> is running uLaw as the codec. Anything else will produce ever
&g
Make sure that if you're using anything other than zaptel hardware, it
is running uLaw as the codec. Anything else will produce ever
increasing delays.
My setup has all of our VoIP lines coming into my main box, and then I
have a separate box running asterisk only for meetme with an iax2 trunk
be
No one has any idea? Even a NO it can’t
be done would be appreciated.
Thanks in advance.
Dan
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Dan Morin
Sent: Monday, June 20, 2005 7:24
PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject
Title: Normal
If someone has a minute, I would appreciate their help
configuring my dialplan. I am using 2 Sipura-2000s to connect to the CO
ports on my legacy PBX. I’m tyring to setup the dialplan so that
when someone enters an extension (1XX), it will determine which of the 4 sip
extens
Asterisk only runs on 5060/udp.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Moises
Silva
Sent: Monday, June 20, 2005 7:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] service scan
i want to make script in
In the queues.conf file, under your queue
you can add the following:
member=sip/ExtensionNumber
where ExtensionNumber is the extension.
Then they should always be part of the queue.
Hope this helps.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf O
Does anyone know if it is possible to use the following disconnect
tone setting with an x100p card?
Disconnect Tone: [EMAIL PROTECTED],[EMAIL PROTECTED];4(.25/.25/1+2)
This tone was written for a Sipura SPA-3000 for a Panasonic KX-TD1232.
The Panasonic does not support disconnect s
y, June 20, 2005 3:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Panasonic KX-TD1232
On Sun, 19 Jun 2005, Dan Morin wrote:
> If anyone has any experience with a Panasonic KX-TD1232 phone system,
I
> would really like to talk to you for a few minu
If anyone has any experience with a Panasonic KX-TD1232
phone system, I would really like to talk to you for a few minutes.
I have asterisk connected to a Panasonic system via FXS
-> CO ports. I’m trying to get the Panasonic configured so that
if someone dials a number (9) while Interc
Yes, I have both nat=yes and canreinvite=no. I'm running version
1.0.6.2 firmware in the budgetone, I upgraded to the newest version
thinking they may have fixed some problems. I've tried it with and
without STUN.
I noticed something very interesting today. Although it can not
register, I can c
Title: Normal
I have a couple of Budgetones that I am playing with
trying to get them to work with * from a remote network over the Internet (yes
NAT joy!). My * server is in my DMZ and I have 5060 and my RTP range forwarded
(UDP) to my public address (through a Cisco PIX). Internally, I
ndex.php?page=Asterisk%20app_conference
>
> I believe it does what you want to do, but I really don't know if it
works
> with CVS_HEAD or stable releases. I'd be curious to hear how it
affects
> performance as well.
>
> MATT---
>
> -Original Message
Title: Normal
So no one has any ideas about how to get MeetMe
to work with a codec other than ulaw?
Is anyone successfully doing it?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Morin
Sent: Tuesday, May 03, 2005 10:26
PM
To: Asterisk Users Mailing List
Yeah, so I’m an idiot…subject
should have been ‘MeetMe’ not MOH.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Morin
Sent: Tuesday, May 03, 2005 10:26
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] MOH Core
uses
I’m trying to get Asterisk setup as a conference
bridge. When I originally tried MeetMe, I was using GSM and as the conference
got longer, the delay got worse and worse. From my research, I assumed that it
was because MeetMe uses ulaw at its core, so everything is getting transcoded
twice
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Morin
Sent: Thursday, April 28, 2005 11:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom IP500 - Phone TIme
>> Wiley Siler wrote:
>&
>> Wiley Siler wrote:
>>
>>> Does anyoe know where I can set the timezone in the configuration
files?
>>>
>>> I am in Phoenix, AZ which has a GMT offset of -7 hours but when I
enter
>>> this into the gmt fields in ipmid.cfg nothing seems to happen.
>>>
>>> Here are the fields...
>>> tcpIpApp.sntp.
;ll have to look in the Panasonic book to see if it
has anything to say on the subject...
Brian Leyton
IT Manager
Commercial Petroleum Equipment
From: Dan Morin [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 27, 2005
12:07 PM
To: asterisk-users@lists.digium.com
Subject: [As
Title: Normal
My company has an old Panasonic KX-TD1232 phone system
that they are using. I want to interface my Asterisk box with this system for
a good conferencing solution. I have two X100P clone cards in my server for
testing. They are hooked up to the analogue phone ports on the ba
I just got a few Polycom IP500s and I’ve been
following the info in the wiki trying to configure them. From what I can
tell, they seem to be setup correctly (well…they don’t work so
obviously not…) however, when they try to register with Asterisk, the following
error shows up in the Logs:
Sorry for the double
post, I tried to paste and accidently sent the email
I've been playing with Asterisk for a few weeks now,
and I've gotten everything to work well with softphones, so I'm ready to move on
to normal VoIP phones. I've been looking around and reading comments that
people
I've been playing
with Asterisk for a few weeks now, and I've gotten everything to work well with
softphones, so I'm ready to move on to normal VoIP phones. I've been
looking around and reading comments that people have had, and I was convinced
that the Polycom IP300 was a great phone for a
Title: Normal
Does anyone know if the X100P clone cards provide the
timer needed to run MOH and the Conferencing service? I have no need for a T1
card, but I’m running asterisk on a dual processor machine with the wrong
kind of USB devices, so none of the dummy timers will work for me.
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