Does anybody know a "type 102 milliwatt test number" that I can dial in the
USA? I need this in order to configure my "rxgain" and "txgain". My analog
line provider, AT&T Repair Center was so confuse, when called them. Thanks in
advance.
-John
__
In my setup, I am using TDM Wildcards analog connections and the Asterisk PBX
box does the converting to my SIP Phones. I had similar problem, when Asterisk
could not recognize my DTMF tones, so I had to tune the FXO modules. Here is
the link to the page:
http://www.voip-info.org/wiki/view/A
I have the SPA962 with the SPA932 side car and it works great, once I got it to
configure correctly for my Asterisk PBX system, which has a Digium TDM03B WILD
CARD installed. I like the Cisco/Linksys SPA models because they have good and
easy configuration webpages for the VOIP phones.
Date:
I had a similar problem when using the TDM03B card with 3 fxo module. In my
cas,e the issue stemmed from a noisy analog line from AT&T, so I had to tune my
TDM card by using fxotune utility. I hope this helps. Check this link out:
http://www.voip-info.org/wiki/view/Asterisk+fxotune
-John
>
Asterisk Users,
I am running a Debian "Etch" system with Asterisk 1.4.11 with a TDM03B card.
Once in awhile, I get this error on the Asterisk, which causes my channels to
be busy/congested, leaving me with just one channel to recieve and make calls:
NOTICE[31454]: chan_zap.c:6367 ss_thread:
I have a TDM04B card and not seen this issue. You may need to check your dial
plan.
> Date: Sat, 12 Apr 2008 14:00:49 -0600
> From: [EMAIL PROTECTED]
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] problem TDM01B
>
> hI list, I have some problems with a TDM01B , when I am tal
Asterisk Users,
I am running Asterisk 1.4.11, Zaptel
1.4.5.1, and Librpi 1.4.1 on a Debian "Etch" system. On the recorded
voice mail messages, the volume is really low when retrieving them with
my cell phone. I tried with multiple cell phones with the volume level
high and still, the sam
Thanks. I will give this a try.
-John
> From: [EMAIL PROTECTED]
> To: asterisk-users@lists.digium.com
> Date: Wed, 2 Apr 2008 09:29:48 -0700
> Subject: Re: [asterisk-users] Voicemail- Recorded Mesage Low Volume
>
>
> On Apr 1, 2008, at 5:22 PM, [EMAIL PROTECTED]
> wrote:
>
> > Can the vol
Grandview, MO 64030
816-767-5577
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan
Sent: Tuesday, April 01, 2008 2:32
PM
To: Asterisk Users
Subject: Spam:[asterisk-users]
Voicemail- Recorded Mesage Low Volume
Asterisk Users,
I am running Asterisk
Asterisk Users,
I am running Asterisk 1.4.11, Zaptel 1.4.5.1, and Librpi 1.4.1 on a Debian
"Etch" system. On the recorded voice mail messages, the volume is really low
when retrieving them with my cell phone. I tried with multiple cell phones
with the volume level high and still, the same
,
John
> Date: Fri, 28 Mar 2008 09:05:02 -0600
> From: [EMAIL PROTECTED]
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] SPA-962+ SPA-932- blf function
>
> John Meksavan wrote:
> > Asterisk Users,
> >
> > I am running Asterisk 1.4.11 on
side-car buttons?
John Meksavan wrote:
Asterisk Users,
I am running Asterisk 1.4.11 on Debian
"Etch" system with the TDM03B wildcard. I recently purchased a
SPA-962 and SPA-932- the sidecar for our receptionist. After reading
many forum postings on how to configure the side car
Asterisk Users,
I am running Asterisk 1.4.11 on Debian
"Etch" system with the TDM03B wildcard. I recently purchased a
SPA-962 and SPA-932- the sidecar for our receptionist. After reading
many forum postings on how to configure the side car, I uprgraded the
SPA-962 software to 5.1.18(SC) v
Asterisk Users,
I am running Asterisk-1.4.11 on a Debian
"Etch" system. On an occasion, when customer calls into my Asterisk Box, I get
this error messagefrom Asterisk
"CallerID returned with error on channel Zap/3-1" , causing all my zap
channels to be busy. So, I cannot make any calls i
What is your setup, hardware wise?
If you have the digium cards- FXO or FXS, you must make sure you tune them. I
had issues with DTMF's, when I went live with my Asterisk system. Once I tune
them, everything worked great.
Date: Wed, 24 Oct 2007 09:05:35 -0500
From: [EMAIL PROTECTED]
To:
Google
SiSky http://www.yeastar.com/ProductsforAsterisk.asp
Regards,
Alejandro Lengua
On 9/6/07, John Meksavan <[EMAIL PROTECTED]> wrote:
>
> Has anybody ever integrated Skype with Asterisk? If you have, which
> software would you recommend to accomplish such a task? ChanSkype
Ira and Doug,
Thanks for your inputs. It seems like there are so many mixed reviews on
every sip provider. In the past, I have used Broadvoice, Vitelity, and
Teliax. All three have all the same problems- call quality and DTMF Tones.
Some days, it would work perfectly fine, while on other
Asterisk Users,
I am thinking about selecting CALLWITHUS as my sip provider. Has anybody
ever used them? How was the call quality? DTMF Tones issues?
Thanks in advance.
-John
_
Gear up for Halo® 3 with free downloads and a
Has anybody ever integrated Skype with Asterisk? If you have, which
software would you recommend to accomplish such a task? ChanSkype? And how
reliable are the calls? Did the DTMF tones work? Thanks in advance.
_
Discover sweet
Password Issue
Date: Tue, 28 Aug 2007 14:53:26 -0800
John, glad it worked for you. Since you didn't feel you needed a name
or email address, you might as well try just commas as delimiters:
201 => 1234,,
Moj
John Meksavan wrote:
> Mojo,
>
> Thanks for helping me with this i
To test that, try adding
another without a pin number:
199 =>
and see if you then get two of the "variable has bad format" error messages
Moj
John Meksavan wrote:
> Here is my voicemail.conf file:
> [default]
> 200 =>
> 201 => 1234
> 225 => 1234
__
should work fine.
Regards,
AFShin
On 8/28/07, John Meksavan <[EMAIL PROTECTED]> wrote:
>
> Asterisk Users,
>
> I am running Asterisk-1.4.11 with Zaptel-1.4.4 on Debian Etch System
> 2.9.18-4-amd64. A TDM03B is installed on the Debian System.
>
> Every time, I try to cha
Asterisk Users,
I am running Asterisk-1.4.11 with Zaptel-1.4.4 on Debian Etch System
2.9.18-4-amd64. A TDM03B is installed on the Debian System.
Every time, I try to change my voicemail pin via the Sip phone, the
voicemail.conf does not get modify and I see this warning message on the
Ast
ers Mailing List - Non-Commercial
>Discussion"
>Subject: Re: [asterisk-users] Experimenting- Sip dialing with Zap
>Date: Thu, 16 Aug 2007 13:00:34 -0500
>
>On 8/16/07, John Meksavan <[EMAIL PROTECTED]> wrote:
> >
> > line yet. The phone simulator only a
=no
canreinvite=no
>From: "James FitzGibbon" <[EMAIL PROTECTED]>
>Reply-To: Asterisk Users Mailing List - Non-Commercial
>Discussion
>To: "Asterisk Users Mailing List - Non-Commercial
>Discussion"
>Subject: Re: [asterisk-users] Experimenting- Sip dial
Asterisk Users,
I have 3 FXO modules with the TDM400P Digium Card. I can dial into the
Asterisk rings my Sip phone, but dialing out with my SPA941 phone through
the zap channel is a problem. I keep getting this message on the Asterisk
CLI. What am I doing wrong? Thanks in advance.
-- Exe
terisk Users Mailing List - Non-Commercial
Discussion
To: "Asterisk Users Mailing List - Non-Commercial
Discussion"
Subject: Re: [asterisk-users] FXO Modules and Sip Outbound
Date: Mon, 13 Aug 2007 16:36:08 -0500
On 8/13/07, John Meksavan <[EMAIL PROTECTED]> wrote:
> Asterisk Users,
>
Asterisk Users,
I have never done a dial plan for this scenario before. Is it possible to
have Sip Phones make outbound calls through the PSTN? What would the call
routing/dial plan would look like?
-John
_
Messenger Café
Asterisk Users,
I am looking for Sip Providers for my Asterisk 1.2.13, running Debian Etch
system with McLeodUSA's T1 service.
Has anybody ever used Callcentric for their Sip Provider? Any service
issues with Callcentric?
Best Regards,
John
_
Asterisk Users,
The more I work with the Asterisk 1.2.13 on the Debian Etch, the more
realize there is no real reliable SIP provider. Having two Sip Providers is
smartest thing to do, one being your main provider, while the other being
the failover/safety.
Ideally, I would like it to fail
r, then:
>
>apt-get install mtr-tiny
>
>and you soon will have :)
>
>(mtr-tiny which is the text/curses version - the 'full' mtr is a GTK
>application, and running X applications on your asterisk box probably
>isn't what you want to do! Mtr *really* doesn&#
Asterisk Users,
I am running Asterisk 1.2.13 on Debian Etch with McLeodUSA's T1 service.
I have two Netgear switches on my T1 router, one for VOIP and another for
data.
I use a gigabit switch for all VOIP and a regular 10/100Mbps switch for
all data. This morning I saw this message a few
ginal Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan
Sent: Wednesday, August 08, 2007 12:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] VoicePulse Connect
Asterisk Users,
Has anybody use Voicepulse Connect for Asterisk?
I am trying
Asterisk Users,
Has anybody use Voicepulse Connect for Asterisk?
I am trying to cover all my bases because in the past, I got burned with
poor quality of service, along with failed DTMF tones with 3 different SIP
Providers (Vitelity, Broadvoice, and Teliax).
I am running Asterisk 1.2.13
quot; <[EMAIL PROTECTED]>
>Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List -
>Non-Commercial Discussion
>To: asterisk-users@lists.digium.com
>Subject: Re: [asterisk-users] Teliax Quality of Service
>Date: Thu, 2 Aug 2007 16:40:27 -0400
>
> On 8/2/07, John Meksavan
Asterisk Users,
I recently ran into some problems with the quality of service with Teliax.
This occurred on August 1, 2007 with a dropped outbound call, audio
quality isse on the callee side- not hearing me well on callee side, and
sending DTMF tones (configured for RFC2833). Am I the only
Users Mailing List - Non-Commercial
Discussion
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Asterisk DTMF Tones
Date: Wed, 1 Aug 2007 15:51:48 -0400 (EDT)
John,
On Wed, 1 Aug 2007, John Meksavan wrote:
> I am running Asterisk 1.2.13 on Debian Lin
>Reply-To: Asterisk Users Mailing List - Non-Commercial
>Discussion
>To: Asterisk Users Mailing List - Non-Commercial
>Discussion
>Subject: Re: [asterisk-users] Asterisk DTMF Tones
>Date: Wed, 1 Aug 2007 15:51:48 -0400 (EDT)
>
>
>John,
>
>On Wed, 1 Aug 2007, John Meksavan
Asterisk Users,
I am running Asterisk 1.2.13 on Debian Linux 2.6.18-4-amd64 and having
problems with DTMF Tones. I have sip service from Teliax and configure to
use rfc2833 for dtmfmode. The problem occurs, when I am using Linksys PAP2T
phone adapter with a regular analog phone.
Is this an
Asterisk Users,
I have Asterisk PBX System running at my work. The system is working
great. Currently, I have Broadvoice as my sip provider and I am not
completely satisfy with their service. Broadvoice only allows 2
simultaneous calls, which hinders my company's communications ability.
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