Hi,
I have just set up 4 7960 with *, they came with V3 f/w, upgraded trough
V5 to V6 fine and they work well, very good quality and flexible feature
set. one of the best phones in my opinion. (Couldnt get V7 firmware on
though just yet)
M
On Fri, 2004-07-02 at 18:26, Chris Glover wrote:
Hi,
On 06:41 PM 7/2/2004, Andrew Kohlsmith wrote:
On Friday 02 July 2004 21:25, Chris A. Icide wrote:
I still didn't get an answer to the original question of will DACS do this?
To my knowlege, DACS is simply a cross-connect.
dacs=1-24:48
When you pick up line 1, * automagically bridges it to line
At 23:48 02/07/2004 +0200, you wrote:
Hi there
I am pretty close on giving up on Asterisk :-/
I am (still) trying to make a call from a H323 phone to an Asterisk
provider using AIX. But H323 does not route the number to AIX. All it is
transmitting is an s.
*CLI -- Executing Dial(OH323/R27865,
Weee, thank you so much for that help.
I can now make calls from my Cisco Call Manager to Musimi :-)
/Martin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Soren Rathje
Sent: Saturday, July 03, 2004 3:33 AM
To: [EMAIL PROTECTED]
Subject: Re:
Im having trouble working out the supported
functionality in Asterisk for Snom 200 extension
monitoring. By setting one of the Snom keys to destination
and entering the extension number I can quick dial the extension by pressing
the softkey. However no status of the extension seems
to be
S
Does Asterisk support pager notification of new voicemails out of the
box? Or do I need an AGI script to do that?
AGI isn't the route for this. Most pagers support an email gateway, just
use it. Maybe you need to trigger it with a procmail rule.
I was asking because my customer
is there a way to do SayDigits() or equivalent that is
backgrounded?
application is
exten = s,1,Background(zz-fwd-areyouat) ; use callerid or enter
exten = s,2,SayDigits(${CALLERIDNUM}) ; telling callerid
exten = _*,1,Macro(fwd-set,${userid},${CALLERIDNUM})
exten =
I have the following situation.
My Asterisk Box is behind firewall ( for example 10.1.1.2 ) I have
mapped 5060,1-10010 and
in rtp.conf I have said this range of prots 1-10010. I'm tring to
dial a PSTN from another PC with Sip phone in internet with external ip.
I can hear the voice from
When I was trying to run mutiple E1s over TDMoE, the zaptel would
drivers complain about too little memory, whenever I added more than 31
channels. Requesting 62 channels in a dynamic span gave me
... span creation failed: Cannot allocate memory
upon loading the zaptel drivers.
How would you go
Thilo Salmon wrote:
How would you go about running, 8 or 16 say, E1s over TDMoE? Would you
create multiple dynamic spans or just one large one? How would you
assign d channels to spans, if you had just one large span?
I would be surprised if the drivers supported more than 31 channels in a
single
On Sat, 2004-07-03 at 19:51, Kevin P. Fleming wrote:
I would be surprised if the drivers supported more than 31 channels in a
single span; these are just virtual PRI spans after all, so there's no
reason for the driver to support an unlimited number of channels per span.
I see. Then the
I can't seem to get music on hold working, it tries to work, but I
just hear strange noises on the extension.. Here is some debug info.
Looks like mpg123 starts fine, but I hear nothing.
I'm on todays CVS build.
-- Executing Answer(SIP/2203-062c, ) in new stack
-- Executing
You want to use mpg123 0.59r, not 0.59q.
-Original Message-
From: Brian Weaver [mailto:[EMAIL PROTECTED]
Sent: Saturday, July 03, 2004 1:23 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Music on hold problem
I can't seem to get music on hold working, it tries to work,
Ok lets see if we can get this in the google search better:
Keywords:
mpg123 asterisk music on hold music-on-hold MusicOnHold
wont start
is slow
is garbled
will not start
gives an error
NOTICE: USE mpg123 0.59r ONLY
http://www.mpg123.de/
bkw
- Original Message -
From: Jay
On Sat, 2004-07-03 at 13:19, Thilo Salmon wrote:
On Sat, 2004-07-03 at 19:51, Kevin P. Fleming wrote:
I would be surprised if the drivers supported more than 31 channels in a
single span; these are just virtual PRI spans after all, so there's no
reason for the driver to support an
On Sat, 2004-07-03 at 11:14, Chris Travers wrote:
S
Does Asterisk support pager notification of new voicemails out of the
box? Or do I need an AGI script to do that?
AGI isn't the route for this. Most pagers support an email gateway, just
use it. Maybe you need to trigger it with
Hi,
Anyone know is there a limitation on internal asterisk database size?
Ta
SJ
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Is there any way for me to add myself to a call queue from outside of my
Asterisk Box?
For example,
I have a queue set up on my asterisk box, and I want to call it on my Cell
Phone, then add myself to the queue and hang up.. When a call comes into the
queue, I want it to be forwarded to my cell
I am trying to receive both CID and DNIS from the telco through a
non-pri T1. Currently I have the T1 setup and operational both outbound
and inbound calls are completed as should be expected. The calls came
in and were placed in the context specified in zapata.conf on exten =
s,1.
I have
On Sat, Jul 03, 2004 at 06:45:13PM -0600, Jared Mashburn wrote:
Is there any way for me to add myself to a call queue from outside of my
Asterisk Box?
For example,
I have a queue set up on my asterisk box, and I want to call it on my Cell
Phone, then add myself to the queue and hang up..
Is there any way for me to add myself to a call queue from outside of my
Asterisk Box?
For example,
I have a queue set up on my asterisk box, and I want to call it on my Cell
Phone, then add myself to the queue and hang up.. When a call comes into
the
queue, I want it to be forwarded to my
I have TE405P span connected
to a patton/digi 2977 T1/PRI port. Zaptel starts up fine without any errors. Libpri
is installed as well.
But I get the following when
trying to load asterisk
---
Jul 3 19:30:26
ERROR[8192]: Signalling requested is PRI Signalling but line is in
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