We have Cisco 9760 for executives and Aastra 9112i for everybody else.
We started with Grandstream, don't remember the model, cost around $80
USD but it had bad audio quality and echo problems (running asterisk
1.09). The quality of construction felt poor, like a toy phone.
We replaced them
810 is an area code in Michigan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Doug
Crompton
Sent: Tuesday, November 14, 2006 10:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 900 rules
Ok so ONLY 900
PROTECTED] Behalf Of Armin
Schindler
Sent: Wednesday, October 18, 2006 3:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CAPI channel not available but nobody is
usingthe system
On Tue, 17 Oct 2006, Tim Sharp wrote:
I have 23 CAPI channels defined
I have 23 CAPI channels defined and normally multiple channels are in use
during the day for outbound calling. The problem is that every 3 or 4 months
one of the channels becomes unavailable and then no calls can come in or go out
on any of these channels. CAPI INFO shows Contr1: 23 B
I am currently running 1.2.7.1 and it works just fine. I personally like to
stay 3 or 4 months behind the current release. This time it is a bit longer
because I don't feel comfortable with the stability of later releases.
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I am
looking at CTPX's VP2000 product. I haven't tried it
yet.
Please
let me know if you find a solution that works.
Tim
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Jonn R
TaylorSent: Friday, September 01, 2006 12:15 PMTo:
I am trying to configure Eicon's SIP version of SoftIP to work with Asterisk.
The documentation is poor and my general knowledge of SIP communications is
limited. I am getting a circuits busy message when trying to call the IP
address of the server with SoftIP. If anybody has gotten this to
Hello,
I get this error message when trying to route an incoming fax from a packet
based T1 to an EICON board that is connected to an external fax voice mail
server.
Voice calls route to this external server with no error. Both fax and voice
calls that come in a channelized T1 also route
I also had an intermittent problem, on average one or two faxes a week, that
were not recongnzed as a fax. Then I switched phone companies and have not had
that problem since. It has been over 2 months. I addition, my echo problem
has been practically eliminated and overall voice quality is
Steven,
I am in Livonia. Please let me know if there is interest.
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of BerkHolz,
Steven
Sent: Thursday, June 22, 2006 4:27 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SE Michigan asterisk
The number dialed after Background is stored in the EXTEN variable and can be
used in the Dial application.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Don
Sent: Friday, June 23, 2006 5:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
The options are not seperated by commas.
exten = s,1,Dial(SIP/50,23,r,d)
should be
exten = s,1,Dial(SIP/50,23,rd)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Klimek
Sent: Thursday, June 22, 2006 2:59 PM
To: Asterisk Users Mailing List -
I am running on 1.2.7.1 and have an intermittent problem when making outgoing
calls. Sometimes the calling party does not hear the ring tone in their
handset, but the call goes through. From my extension I have only had 3 calls
like this in the last couple of weeks, other people have had 3 or
tell you to use the r option
to Dial. Those people are confused. Don't do that until you have tried
everything else.
Tim Sharp wrote:
I am running on 1.2.7.1 and have an intermittent problem when making outgoing
calls. Sometimes the calling party does not hear the ring tone
Yes it does, I just set our system up that way.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Gareth
Blades
Sent: Tuesday, June 06, 2006 6:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Playback welcome
I am
running on 1.2.7.1
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of
OlivierSent: Tuesday, June 06, 2006 12:17 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Playback welcome message while
FYI, we upgraded to 1.2.7.1 this weekend and the new features in MOH allow for
the sound file to start at the begining with each call.
Thanks to Chris and Matt for their responses.
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tim Sharp
Sent: Friday
-Users] Music on Hold restart at beginning for
each call
On Tue, 16 May 2006, Tim Sharp [EMAIL PROTECTED] wrote
Chris,
When I tried background it waited until the message was done before
dialing, just like playback. Am I missing something?
Wasn't my suggestion :)
If I've understood what you're
List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Music on Hold restart at beginning for
each call
Why not use background?
On 5/15/06, Tim Sharp [EMAIL PROTECTED] wrote:
Chris,
I have it working now using Playback but I am looking for a way to reduce the
wait time.
The system
.
Thanks,
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris
Hastie
Sent: Sunday, May 14, 2006 5:37 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Music on Hold restart at beginning for
each call
On Fri, 12 May 2006, Tim Sharp
I am using the m option on the dial command to play a message instead of
ringing. The message is something like please wait while I try to locate your
party so I need it to start at the beginning for each call. I think there
might be a way in 1.2.x be we are not ready to upgrade yet so a
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