Doesn't seem to help. I did it early yesterday morning and have
another 'stuck' call this morning
Does anyone have any other ideas on what I can do to correct this?
thanks
Shawn
CLI core show channels
Channel Location State Application(Data)
DAHDI/8-1
Right, this is how I expected it to operate. My prior question though was
regarding the 'T1 over Ethernet' scheme someone mentioned which ran full
throughput all the time.
That is true. If you're doing a clear-channel or pseudo-wire T1 over
ethernet you will always be using 1.54 Mbps
I have a situation where I have an Asterisk box which receives 8
analog lines from a
Mitel PBX and then drives 8 cordless SIP phones in a 1-to-1 mapping (a
call coming in
on port 1 of the digium FXO board is delivered to SIP phone 1, an
outgoing call on SIP
phone 2 goes out FXO line 2, etc.
This
I'm setting up an asterisk server to extend several extensions from a mitel pbx.
I'd like to display the caller id that I receive from t he mitel pbx
on the sip phone. The mitel
PBX person has setup the PBX to send be callerid, but I don't see it.
I've set chan_dahdi up with
usecallerid=yes
Yes, I'm talking about mid-call.
I do have rtptimeout and qualify set, both to 30 seconds, which should be
plenty of time.
I set them both because if a phone moves out of range, and never comes back,
asterisk was keeping the channel open way to long.
On Wed, May 4, 2011 at 7:50 PM, Matt Riddell
I have a situation where we have an asterisk box that is extending several
Mitel PBX extensions to
some cordless SIP phones (Cisco WIP310). Everything works great, except
when the cordless
phone walks out of range of one access point and into range of another
(cisco 1100 series APs).
I've been
I have a situation where we have an asterisk box that is extending several
Mitel PBX extensions to
some cordless SIP phones (Cisco WIP310). Everything works great, except
when the cordless
phone walks out of range of one access point and into range of another
(cisco 1100 series APs).
I have
I have 2 separate Asterisk servers that are both exibiting this problem. 1
has a 4 port
FXO digium card, the other an 8 port.
For some reason when the machine reboots, the dahdi drivers are not properly
loaded. Then asterisk
ends up starting without dahdi support. I've tried everything that I
This one is a little off-topic, it's more about the phone than asterisk
itself.
I have a cisco 7960 configured with 2 lines to 2 different sip providers
(cant get
asterisk to register with the 2nd provider, but that's another story). Is
there a
way yo determine which direction speed-dial buttons
Hello List,
Does anyone know of a device that works with the TDM400P FXO/FXS Modules to
provide line backup for power failure?
I have an idea for such a device but do not have enough vision to do the
soldering of the parts.
My ideal device would be able to sense when asterisk has brought the
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Power Failure, Line Switch, Relay device
On Mon, 2005-01-10 at 14:48 -0600, Shawn L. Djernes wrote:
Hello List,
Does anyone know of a device that works with the TDM400P FXO/FXS Modules
to
provide line backup
Michael
Yes is works simular to the Zap stuff
You put context=whateverIncomingContext in the sip.conf block for the
device or port. Like I have everything that comes in on SIP go into the
context sipfriend execpt my 2 ata ports and the 2 lines of the 7940
phone.
Looks something like this.
Hi,
I have Zaptel on Debian Testing with 2.4.24 kernel. I am building stuff
from CVS instead of using packages for right now. I like to get my bug
fixes fresh everyday or 3.
Here is what I did to make it work.
Had Deb testing already running.
Built 2.4.18,19,20,22,24 (had zap working with all
Do you here the beeps on the phone or on the Console machine. For about the
last 2 weeks I have been hearing random beeps on either of my two sip
phones. I do not have a console running anywhere so I have no text
printing.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
From What I understand of this feature it is only to keep the phone working
not to provide full services. I think they intended it to be something like
a less powerful router or a box at a remote site. This way if the primary
server was took out by a virus or hardware failure your office staff
equipment to reset
around here.
Thanks
---
Shawn L. Djernes,
N0PEU
Djernes.org
[EMAIL PROTECTED] | [EMAIL PROTECTED] | [EMAIL PROTECTED]
http://www.djernes.org
PH: (412)-257-3252
519 Washington Ave. #2, Bridgeville, PA, 15017,
USA
. When your mailer adjusts the
size to smaller than normal, it becomes a problem.
On Sun, 2003-07-13 at 13:16, Shawn L. Djernes wrote:
Hello,
I am trying to solve a problem that I can foresee when I deploy
Asterisk into a few SOHO situations soon. In Nebraska and in my area
of Western
Hello,
If you
have the SIP firmware load in the unit then UseSIIP should be the first
yellow box in the left column. If
you do not have that you may have a MGCP version and need to get a hold of the
SIP firmware and Flash the unit first.
Shawn L.
Djernes
-Original
Message
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Florian Overkamp
Sent: Friday, June 27, 2003 14:02
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RE: [Asterisk-Users] cisco 186 helpp!ª
Citeren Shawn L. Djernes [EMAIL PROTECTED]:
Hello,
If you have the SIP firmware load in the unit
From Reading http://www.cisco.com/univercd/cc/td/doc/pcat/827.htm that
device is H323 and only has FXS ports. So any of the Cisco Docs on H323
would probably point you in the right direction.
Shawn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dave
20 matches
Mail list logo
