I've searched
through the newsgroup and online and haven't found an answer for my question...
maybe I am looking for the wrong terms, I am not sure...
I have a client that
would like a phone that is like a "typical" receptionists
phone.
Requirements:
- Ability for
their 3 lines to "ligh
t "static" on the line.
Jeff Busch
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
"ManxPower" Wieling
Sent: Wednesday, December 07, 2005 11:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asteri
dn't be relevant,
correct?
Thanks!
Jeff Busch
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
McMullen
Sent: Tuesday, November 29, 2005 6:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-User
rcial Discussion
Subject: Re: [Asterisk-Users] Static on inside end of conversation
Jeff Busch wrote:
>You are correct. Inside means the IP Network and Outside means the
>PSTN accessible via the Audiocodes MP-108 gateway.
>
>No QoS.
>
>
>
We've seen echo on congested LANs
: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Static on inside end of conversation
Jeff Busch wrote:
>Hello,
>
>I am running the following configuration:
>
>2.8ghz P4 with 1GB of RAM
>Audiocodes MP-108 connected to 5 POTS lines Polycom IP-5
Hello,
I am running the following configuration:
2.8ghz P4 with 1GB of RAM
Audiocodes MP-108 connected to 5 POTS lines
Polycom IP-500 phones
[EMAIL PROTECTED] 1.3 (this is Asterisk 1.0.9)
End users are complaining of an echo and static on the inside end (the
internal side), but the outside end o
found that for these small office
implementations, this has not been an issue.
Hope this helps.
Jeff Busch
[EMAIL PROTECTED]
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
FelderSent: Sunday, July 17, 2005 8:17 PMTo: Asterisk
Users Mailing List - Non-Commercial Di
Login as administrator.
Upper right, click on "admin"
Choose "configure tabs"
You can decide which tabs to show/hide here.
- Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Tuesday, June 07, 2005 11:04 AM
To: David John Walsh;
That was the fix.
For other people who might be having this problem, I am now using:
exten => _8NXXNXX,1,Dial(Zap/g1/ww${EXTEN:1})
Thanks!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Webb
Sent: Friday, March 04, 2005 1:49 PM
To: Asterisk Us
production box on a small construction job site.
Thanks!
Jeff Busch
[EMAIL PROTECTED]
-Original Message-
Subject: [Asterisk-Users] Problems dialing out - possible settings
changes
First, I apologize if this info has been covered, I tried doing a search
on Google and the wiki and found
is for help if you run into an
implementation problem, not to teach you the basics by using 1, 2, & 3.
Thanks for your thoughts on this!
Jeff Busch
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shadow
Roldan
Sent: Thursda
Fixed my own issue after a little more work and testing.
Ended up that I was missing the "context=localextensions" line in my
sip.conf for the new extensions I had setup.
Thanks!
Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff
First, I apologize if this info has been covered, I tried doing a search
on Google and the wiki and found nothing, but I could be searching for
the wrong info.
Second, I am new to asterisk and linux. My knowledge is Asterisk is
much better than linux, so excuse the ignorance in some areas.
Confi
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