Dear Sirs & Madams,
Has anybody connected and sucesffully used a Cisco 7910 to an * PBX
system? Also, how hard was it to setup?
Any assistance would be greatly appreciated.
Kind Regards
Stuart
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://l
I have setup AgentCallbackLogin and the agents have been logged in
successfully.
However when calls are queued and an agent picks up the call. It just
hang up the call.
On the command console it does say the agent "agent 1001 hang up on
customers. they must be pissed off". I agreed.
My queues.conf
http://www-306.ibm.com/e-business/doc/content/lp/prodigy.html?P_Site=S90
Linux, the Future is Open.
An IBM Commercial shown with the Child Prodigy, it's not the first time
they've shown it.
- Brent
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Go
[EMAIL PROTECTED] <> wrote:
> Linux, Shake up the world
>
> oops sorry,
>
> test... testing 123
>
[EMAIL PROTECTED] <> wrote:
> I laughed out loud, and then looked around at all the other people in
> the room who were staring at me because they didn't understand the
> significanc
Title: Message
Yeah,
but without a sound card; they won't work. My suggestion would be to place
a call from an outside line (or cell phone) through the * to voicemail or the
demos to prove that the system works.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PRO
Use a soft phone as an endpoint. There are a
variety of SIP and IAX softphones you can use to place a call through your
Asterisk box over IP.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Zheng
Sent: Sunday, February 01, 2004
7:34 PM
To:
[EMAIL PROTECT
I laughed out loud, and then looked around at all the other people in
the room who were staring at me because they didn't understand the
significance of the statement.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Juan
Cardenas
Sent: Sunday, February 01
Has anyone had any success using the ztdummy module and
doing meetme/conferencing with out zaptel hardware installed?
Paul
At 10:34 PM -0500 2/1/04, John Todd wrote:
At 3:09 PM -0600 2/1/04, Michael Graves wrote:
Anyone here have experience with these devices? They would ppear to be
an affordable alternative to multiple X100Ps.
Michael
--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specialis
At 3:09 PM -0600 2/1/04, Michael Graves wrote:
Anyone here have experience with these devices? They would ppear to be
an affordable alternative to multiple X100Ps.
Michael
--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specialist www.pixelpower.c
Hi, all I only have x100p card, and my pc doesn't have sound card. How can I make a call and accept a call with x100p on my pc through the telephine line? If these tests are passed, I can get more money to buy a TDM 400 card and a sound card, even an IP phone. Any one can help me? I installed Redha
Scott,
You can't see DNIS on any channels/Line? If I understand correctly you can't
see digits coming in from your Telco? And it falls into your S context?
Try changing your Adtran Firmware, I tried on L35 and L36 it didn't worked
properly, so I am using L34.
Also, try changing the wink time in
Linux, Shake up the world
oops sorry,
test... testing 123
On Sun, Feb 01, 2004 at 04:51:30PM -0600, Steven Critchfield wrote:
>
> This isn't intended as a flame bait. The original message should have
> been more clear that I thought you where experiencing crap in windows.
Heh. I haven't used windows since 1995 :)
In fact, with HP-UX you cannot delete o
As far I have reasearched none of the FXO devices were perfect except Cisco
VIC ones.
If you are looking for reliability I recommend not to use those ones except
cisco.
Kannaiyan
- Original Message -
From: "Michael Graves" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, Febru
Hello,
I have 6 analog lines that ring down coming
into my office that support DNIS, my current phone system (SRX) displays
the "called number" on the screen of the operator phone, IE; xxx-7873 = Netxn,
xxx-7874 = Dolphinsafe, etc.
Does asterisk support any type of features to
distingui
> > From what I can tell (box is about 48 hrs old for me), it
> > seems to be a rather incomplete or just-bare-sip-minimum
> > functionality. It also appears as though all four ports are
> > treated as a group-of-lines, and one doesn't have any choice
> > (from a sip perspective) on which port
On Sun, 2004-02-01 at 16:38, William Waites wrote:
> On Sun, Feb 01, 2004 at 04:21:23PM -0600, Steven Critchfield wrote:
> >
> > Dude maybe you need to learn more Unix programing and leave those toy
> > OSes alone. Once a module is loaded, there should be no need to read the
> > version on the fil
On Sun, Feb 01, 2004 at 04:21:23PM -0600, Steven Critchfield wrote:
>
> Dude maybe you need to learn more Unix programing and leave those toy
> OSes alone. Once a module is loaded, there should be no need to read the
> version on the file system again. Your problem would be loading new
> modules i
On Sat, 2004-01-31 at 20:02, William Waites wrote:
> On Sat, Jan 31, 2004 at 07:43:46PM -0600, Brian West wrote:
> > Nope I do make install all the time with asterisk running without ONE
> > problem.
>
> As I said, this behaviour is specific to some implementations
> of dynamic loadable modules. I
Hi
> Can I define a variable in globals like this:
>
> [globals]
> timeout=60
>
> and then in another context, redefine that same variable and only have
> the new value affect the call that hit that particular extension ?
yes, works this way.
mind : each var is unique for each call... a global
v
> > I don't believe the above will work. There is only one IP address for
> > the box, and no way that I've found to send a sip packet to the box with
> > "additional" information that would suggest using port 1 vs port 2. From
> > what others have hinted at (and it seems the majority of us are lim
[EMAIL PROTECTED] wrote:
I'm trying to set up an Asterisk system for a small office, and one thing
I haven't figured out yet is how to best provide reliability. One way
to go seems to be a T1 for all my incoming phone lines. What if that T1
goes down? Can I use mutiple POTS lines in conjuction
Can I define a variable in globals like this:
[globals]
timeout=60
and then in another context, redefine that same variable and only have
the new value affect the call that hit that particular extension ?
[example]
exten => _9NXX,1,DBget(blah/blah)
exten => _9NXX,102,Goto(3)
exten => _9
Anyone here have experience with these devices? They would ppear to be
an affordable alternative to multiple X100Ps.
Michael
--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc.
On Sun, 1 Feb 2004 08:21:55 -0600, Rich Adamson wrote
[long snip]
> No, the manual is very verbose but no * examples at all. The
> box sells as either a 323 or sip, with different images
> (sort of like C7960's) and different manuals.
>
> The box does not support the "register" function in ei
I'm trying to set up an Asterisk system for a small office, and one thing
I haven't figured out yet is how to best provide reliability. One way
to go seems to be a T1 for all my incoming phone lines. What if that T1
goes down? Can I use mutiple POTS lines in conjuction with a T1, all
connecting
On Sun, 1 Feb 2004, Rich Adamson wrote:
> I don't believe the above will work. There is only one IP address for
> the box, and no way that I've found to send a sip packet to the box with
> "additional" information that would suggest using port 1 vs port 2. From
> what others have hinted at (and it
Rich Adamson wrote:
Product Review
Mediatrix 1204 4-Port SIP FXO Gateway
Firmware: v2.4.10.69 - US Version
US Retail: ~$750, Street Price: ~$450.
Trouble shooting is limited to the SNMP manager only. The manager can be used
to view configuration data, however needed dynamic operational statistics
Greg,
> > So try something like this in extensions.conf:
> > exten => 101,1,Dial(SIP/@mediatrixport1)
> > exten => 102,1,Dial(SIP/@mediatrixport2)
> > exten => 103,1,Dial(SIP/@mediatrixport3)
> > exten => 104,1,Dial(SIP/@mediatrixport4)
>
> Oops, maybe I should have written these extensions to be
I did get it to work, and can place and receive calls through the Firefly
network via *.
Compared to iaxtel or FWD, there is a significantly higher amount of
latency, but it is workable.
For some reason, this needed to be the last entry in my iax.conf or it
would try to authenticate with a diffe
On Sun, 1 Feb 2004, Greg Hill wrote:
> So try something like this in extensions.conf:
> exten => 101,1,Dial(SIP/@mediatrixport1)
> exten => 102,1,Dial(SIP/@mediatrixport2)
> exten => 103,1,Dial(SIP/@mediatrixport3)
> exten => 104,1,Dial(SIP/@mediatrixport4)
Oops, maybe I should have written these
Product Review
Mediatrix 1204 4-Port SIP FXO Gateway
Firmware: v2.4.10.69 - US Version
US Retail: ~$750, Street Price: ~$450.
The Mediatrix 1204 SIP FXO gateway is equipped with four RJ11 pstn jacks
and one RJ45 Ethernet jack on its rear panel. It terminates the four pstn
lines in either Loop Sta
On Sun, 1 Feb 2004, Rich Adamson wrote:
> The above does not seem to work either. Since the mediatrix has four pstn
> ports, there must be a way to construct a Dial command that would embed
> a userid:password, port alias name, or something like that. Just can't find
> any reference to what that sy
I haven't taken the time to reverse engineer this on * but subscribe is
used in SIP for serveral things:
1. Message Waiting Indicator (MWI). Asterisk seems to send out a NOTIFY
even with no SUBSCRIBE
though. :)
2. SIMPLE (SIP Instant Message & Presence Leverage Extensions). The
SUBSCRIBE/NO
Chris Craft wrote:
On Saturday 31 January 2004 21:31, you wrote:
<<>>
I am just a low level c hack. Before I go out and write any thing to do
this snmp admin stuff,
are there any linux tools I could use to do this?
Net-SNMP (http://freshmeat.net/projects/net-snmp/ , formerly UCB-SNMP or
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
> Sent: Saturday, January 31, 2004 10:05 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] PCI expansion slots.
>
>
>
> Hello,
>
> Did anyone use PCI expansion slots such as:
>
> http
Hi, all
I only have x100p card, and my pc doesn't have sound card. How can I make a call and accept a call with x100p on my pc through the telephine line? If these tests are passed, I can get more money to buy a TDM 400 card and a sound card, even an IP phone. Any one can help me? I installed Red
Hello.
I'm a bit puzzled at the moment. I have a x100P and TDM400 with 4 modules
(extensions). Asterisk CVS-02/01/04-06:55:30
Part of my extensions.conf says this:
; Zap Phone #1
;
exten => 204,1,Dial(Zap/2,20) ; Ring for 20 seconds
exten => 204,2,Voicemail(u${EXTEN})
exten => 204,3
Thanks, the problem was with crc. It is ok now.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C. Maj
Sent: Sunday, February 01, 2004 12:31 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] TE410P E1 PRI problem
On Sat, 31 Jan 2004, Tomica Crnek w
anyone seen or used * voicemail on a PBX system that needs to talk SMDI
to the VM host?
Dave Packham
Dave Packham
University of Utah Netcom
Campus R&D
c. [EMAIL PROTECTED]
w. [EMAIL PROTECTED]
[EMAIL PROTECTED]
Trillian/ICQ#:45818442
MSN [EMAIL PROTECTED]
Our Groups Website
http://r
> >>>What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
> >>>
> >>>Been trying stuff similar to:
> >>>exten => _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
> >>>where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
> >>>even try the IP.
> >>>
> >>>Rich
> >>>
>
Rich Adamson wrote:
What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
Been trying stuff similar to:
exten => _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
even try the IP.
Rich
from my extensions.conf:
> >What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
> >
> >Been trying stuff similar to:
> > exten => _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
> >where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
> >even try the IP.
> >
> >Rich
> >
> from my extensio
Florian Overkamp wrote:
Hi,
-Original Message-
So, what hardware or use is the SUBSCRIBE method used for in
chan_sip.c? I asked this question a while ago, and got
resounding silence. Maybe someone who is better at
de-tangling C code than I am could take a peek.
Hmm, dunno. Could i
Mike,
I'm hoping one can specify a particular mediatrix "port" in the Dial Sip
command, but haven't found any Dial syntax that would allow passing a
userid/password to the gateway. Since the 1204 provides a AuthUsrPwd on
a per port basis, my guess would be that we either have to pass the Alias
defi
Florian Overkamp wrote:
Hi,
-Original Message-
So, what hardware or use is the SUBSCRIBE method used for in
chan_sip.c? I asked this question a while ago, and got
resounding silence. Maybe someone who is better at
de-tangling C code than I am could take a peek.
Hmm, dunno. Could i
Hi,
> -Original Message-
> So, what hardware or use is the SUBSCRIBE method used for in
> chan_sip.c? I asked this question a while ago, and got
> resounding silence. Maybe someone who is better at
> de-tangling C code than I am could take a peek.
Hmm, dunno. Could it be used to have
Rich Adamson wrote:
So, what hardware or use is the SUBSCRIBE method used for in
chan_sip.c? I asked this question a while ago, and got resounding
silence. Maybe someone who is better at de-tangling C code than I am
could take a peek.
Not sure, but seems to me it came in about the time Olle
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