No problem. First you need to know if the problem is that the
CallManager is not sending anything or if Asterisk is not handling the
conection. You can use tcpdump or ethereal for that.
Salu2
Andrés
Chad Whitten escribió:
Mind sharing how you got asterisk working with callmanager as an h323
I guess the question to ask is... Is the macro function designed to
execute one extension logic and then exit back to it's original context,
or is it designed to allow you to run multiple extension logics before
kicking back?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Nate Carlson wrote:
Hey all,
I've got a PRI used for data calls right now, terminated in a MAX 4000.
We're only using around 12 channels on average and 16 max, so I'd like to
split off the remaining channels to terminate in an Asterisk box.
Does anyone know of a device that'll take a PRI in, and
firmware WF.00.11 is the latest. That's probably what you have. Let me
know it it's not.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, August 10, 2004 11:16
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Pulver wisip
John Todd wrote:
At 7:14 PM +0200 on 8/10/04, Soren Rathje wrote:
Gang,
[snip]
/Soren
It is the mark of an educated mind to be able to entertain a thought
without accepting it.
- Aristotle
Ok, so we moved here from *-dev, no problem... ;-)
VOIP Spam is actually pretty trivial to
hank [EMAIL PROTECTED] wrote:
voip spam?
I have never gotten any yet.
It's is just waiting for the first one to arrive..
The mechanics are just too appealing for spam-like businesses.
Imagine a telemarketeer script that dials lists of VoIP addresses. Instead of having
to pay for each call
Sean Cheesman wrote:
I guess the question to ask is... Is the macro function designed to
execute one extension logic and then exit back to it's original context,
or is it designed to allow you to run multiple extension logics before
kicking back?
If I can do a Goto inside a macro, isn't that
Hello
All,
I have Polycom IP
500 phones which I would like to have message waiting indicators on. So
far, I have my system running well but the problem I am seeing is that MWI
doesn't seem to tell my phone that it should display a MWI state. The
light does not show when you have message
On Tue, 10 Aug 2004, Michael Welter wrote:
Vina Integrator 300.
Looking at the product specs, it looks like it can only spit out a single
T1 and a bunch of FXo/FXS ports, not two T1's. Am I missing something
there?
| nate
Okay, this one is driving me nuts.
I have a fedora core 1 machine running asterisk from CVS. Built last
week. I have a couple of snom phones with the latest firmware.
Here's the issue, it's a wierd one.
You start up the phones, they register, all is good. They show up in sip
show peers like
[EMAIL PROTECTED] (Loek Gijben) writes:
hank [EMAIL PROTECTED] wrote:
voip spam?
I have never gotten any yet.
It's is just waiting for the first one to arrive..
The mechanics are just too appealing for spam-like businesses.
I got one the other day, but it turns out it was a buddy trying
On Aug 10, 2004, at 1:14 PM, Loek Gijben wrote:
hank [EMAIL PROTECTED] wrote:
voip spam?
I have never gotten any yet.
It's is just waiting for the first one to arrive..
The mechanics are just too appealing for spam-like businesses.
Imagine a telemarketeer script that dials lists of VoIP addresses.
-Original Message-
From: Wiley E. Siler [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 10, 2004 4:33 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Polycom IP 500 - MWI Not Working
Is tehre anyone out there with Polycom
phones who has Message Waiting Indicators working with the IP
Interesting ... so just the kernel 2.6.7 and make linux26; make
install made it work for you? Or did you have to create the files as
described at
http://voip-info.org/tiki-index.php?page=Linux%20Fedora#comments as well?
Thanks,
Oliver
Jean-Yves Avenard wrote:
Or you can install the kernel
Nate Carlson wrote:
On Tue, 10 Aug 2004, Michael Welter wrote:
Vina Integrator 300.
Looking at the product specs, it looks like it can only spit out a single
T1 and a bunch of FXo/FXS ports, not two T1's. Am I missing something
there?
Correct, it has 2 T-1 ports and FXO (up to 24 ports) but one T-1 is for data
and the other can be a mix of data and voice.
j
- Original Message -
From: Nate Carlson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 10, 2004 2:53 PM
Subject: Re: [Asterisk-Users] Semi-OT:
Just make sure you have defined the mailbox number in your sip.cfg file.
example:
[222]
context=local
type=friend
host=dynamic
username=222
password=p222
dtmfmode=inband
defaultip=10.1.2.222
mailbox=222
That's all it took for my asterisk + IP500 setup.
Niles
On Aug 10, 2004, at 3:33 PM, Wiley
Okay, time for an update.
I posted this as a bug. Very quickly got informed that it is not a bug,
but instead, an undocumented 'feature'.
AbsoluteTimeout is treated as an *exception* (ie it looks like a hangup)
by most applications, including Macro, which makes most applications exit.
This
The Adtran Atlas 550 or 830 may do what
you're looking to do. We use it to split PRI into multiple BRI.
chris
Nate Carlson [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
08/10/2004 01:30 PM
Please respond to
[EMAIL PROTECTED]
To
[EMAIL PROTECTED]
cc
Subject
[Asterisk-Users]
I have written an application (win32) to view parked calls. The reason
i wrote the code is because my GS doesnt support the system reading the
parked extension back to me after I park it.
This will allow you to view where the call is parked, the caller id and
when it was parked there.
You can
Can someone please tell me which version tags I should be using when
checking out with cvs. Right now I have two directory trees:
1) /u1/src/asterisk/asterisk-stable
2) /u1/src/asterisk/asterisk-dev
#1 has Tv1-0_stable in the CVS/Tags file. I haven't used tags a whole
lot,
On Tue, 10 Aug 2004, Michael Welter wrote:
It has a channel bank function, which you can ignore. It takes one
inbound T-1 integrated PRI from the LEC and splits the data and voice
channels. The data is sent to the LAN (Ethernet) port and the voice
channels are passed to the outbound T-1
On Tue, 10 Aug 2004 [EMAIL PROTECTED] wrote:
The Adtran Atlas 550 or 830 may do what you're looking to do. We use it
to split PRI into multiple BRI.
Yeah, I've been looking into their product offerings, and it does look
like they very likely do what we need. I'll try to find a reseller to tell
Nope. That fixed it. Thank you!
Wiley
-Original Message-
From: Robert Jackson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 10, 2004 2:16 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Polycom IP 500 - MWI Not Working
-Original Message-
From: Wiley E. Siler
Hi,
I've got a recent copy of CVS (09/08/04) with the UK CID patch on and working,
but I'm having trouble receiving SMS messages.
I can send messages without any problems but whenever an incoming message is
detected, nothing is left in the /var/spool/asterisk/sms directory.
Can anyone help?
I saw somewhere that the last kernel to work properly with the zaptel
drivers when using data over it was 2.4.20. Has this been since fixed to
work with newer kernels?
-Mike
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Dan Mahoney, System Admin wrote:
Okay, this one is driving me nuts.
I have a fedora core 1 machine running asterisk from CVS. Built last
week. I have a couple of snom phones with the latest firmware.
Here's the issue, it's a wierd one.
You start up the phones, they register, all is good. They
Hey Folks,
Is it possible to transfer an incoming call back out without a trombone
effect.
For instance;
Caller dials my broadvoice # -- Asterisk Answers and plays a menu -- the
caller selects an option -- asterisk transfers the call to my cell phone
via broadvoice and removes itself from the
- Original Message -
From: Christopher Jacob [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 10, 2004 3:57 PM
Subject: [Asterisk-Users] SIP Transfers (Possibly reinvite)
Hey Folks,
Is it possible to transfer an incoming call back out without a trombone
effect.
For
On Tue, Aug 10, 2004 at 02:12:51PM -0700, Scott Laird said:
On Aug 10, 2004, at 1:14 PM, Loek Gijben wrote:
hank [EMAIL PROTECTED] wrote:
voip spam?
I have never gotten any yet.
It's is just waiting for the first one to arrive..
The mechanics are just too appealing for spam-like
Original Message -
From: Walt Reed [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 10, 2004 4:34 PM
Subject: Re: [Asterisk-Users] Re: VoIP SPAM, what's next ?
On Tue, Aug 10, 2004 at 02:12:51PM -0700, Scott Laird said:
On Aug 10, 2004, at 1:14 PM, Loek Gijben wrote:
On Aug 10, 2004, at 4:34 PM, Walt Reed wrote:
On Tue, Aug 10, 2004 at 02:12:51PM -0700, Scott Laird said:
Why stop there--you can beam pre-recorded messages to phones without a
person or phone line ever being involved. You could send hundreds of
calls per minute without paying for more then a
[EMAIL PROTECTED] wrote:
I saw somewhere that the last kernel to work properly with the zaptel
drivers when using data over it was 2.4.20. Has this been since fixed to
work with newer kernels?
I'm using 2.6.7-gentoo-r13. I don't know where you saw the reference
to 2.4.20, but I've been
-Original Message-
From: Ryan Parlee [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 10, 2004 6:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] CVS version tags
Can someone please tell me which version tags I should be
using when checking out with cvs. Right now I
Guys:
Have you sorted this issues?
I am interested to know if C.A.S. is supported.
Any idea?
I saw googled some posts from the developers, but as far as January this
year, it was still under development.
Much appreciated beforehand,
Carlos H.
Marcelo Rodriguez wrote:
Well I turn on the pri
Greg Hill wrote:
On Mon, 9 Aug 2004, Kevin Johnson wrote:
When dialing 8437624, I get the following output:
-- Executing NoOp(SIP/office1-b727, call for 843762 43762
6) in new stack
on the following line:
exten = _8.,3,NoOp(call for ${EXTEN} ${EXTEN:1} ${LEN(${EXTEN})})
this is really odd.
I posted a comment on how I did it on the exact page you referred to
(click on the *comments* link)
So yes, after I got the 2.6.7 kernel installed (which is very trouble
free once you copied the original fedora .config file)
compile, install , reboot the machine
then compile asterisk normally.
I'm using linux 2.6.7 (FC2 distribution) with zaptel and a TDM403 card
without any problems
Jean-Yves
On 11/08/2004, at 8:41 AM, [EMAIL PROTECTED] wrote:
I saw somewhere that the last kernel to work properly with the zaptel
drivers when using data over it was 2.4.20. Has this been since fixed
I want to have a personal meetme conference, so when
on a call I can transfer the other party to my personal conference with #7.
(I can then make other calls, and dump them into the conference
using #7 as well, then join myself by dialing 7).
Using:
exten = 7,1,MeetMe(${CALLERIDNUM}|Mpd)
this
Soren Rathje wrote:
Next thing will probably be a sbl.e164.org service to block spammers like we do with email... :-)
Actually we don't need to do that, using normal NAPTR record can be used
instead.
We know the IP the call is coming from, we can find out from the NAPTR
where calls normally go to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 10 August 2004 03:30 pm, mattf wrote:
An AGI script can exit any way you want it to. You can have it set the
extension and the Priority to whatever you desire, you're not limited to +1
or +101 you can have it be anything.
As for speed,
hello wolfgang.
I am curious did the packit have any sound to it when your friend tried it
out? I am assuming voip spam would have audio.
thanks
hank
- Original Message -
From: Wolfgang S. Rupprecht [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 10, 2004 2:10 PM
Subject:
On Tue, 10 Aug 2004, Todd Lieberman wrote:
Looks like a firewall issue too me.
Some of the snoms are behind NAT. However, my test one was on the same
subnet, and exhibited the same problems.
The asterisk box has firewalling disabled. A firewall issue, I would
think, would not cause a
Actually, the script I sent him was an AGI script that is part of the freely
available astGUIclient suite http://astguiclient.sf.net/ . Just download the
package and take a look at the .agi scripts in there.
MATT---
-Original Message-
From: Steve Szmidt [mailto:[EMAIL PROTECTED]
Sent:
Hi all,
I am currently running asterisk (CVS HEAD)on
a p4 machine with rh7.3
and using nufone's h323 channel driver.
I was using the old voiceage g729 and have replaced
it with the digiums
g729.
I used to have the message "Measured length exceeds
frame length" but
since I changed the
Shouldn't it be possible to pipe the channels for the MAX through the
Asterisk box ?
The whole PRI into Asterisk and a PRI cable from a second port to the
MAX.
I haven't looked much at data calls from Zap to Zap, but it looked like
it was possible.
Kind regards,
Martin List-Petersen
On Tue,
If you enable the Asterisk transfer using the # key, the grandstream
will read it back to you.
You just have to add a t to the end of your dial statement for your
incomming calls.
On Tue, 10 Aug 2004 15:08:34 -0700, Kyle Hagan [EMAIL PROTECTED] wrote:
I have written an application (win32) to
Chris,
Actually it is a documented feature of Macro. Macro only executes
extension s there are no other extensions in the macro context. I ran
into this while working through building our dial plan. It was driving
me nutz. (But the WIKI rescued me.) What I had to do is use the Macro
to setup
I've searched high and lo and googled to I can't google no more... I
knew that Cisco bought Selsius to get their VoIP solution but what I
didn't know was that the 12sp+ is based upon an ITE-12 product that is
apparently used at universities. I've taken two of the phones apart and
started
With the description you give, the first thing I think of is..
Some of those chips may be EEPROM memories, and they get erased with the
light..well infrared light to be more precise, and only if you remove
the cover stickers I am just guessing though, I have little information
on what actual
Hi,
I was wondering if any CISCO users out there knows if it is possible to
Change the locations of the BUTTONS along the bottom of the screen.
I ask this as the TRANSFER button is only accessible after pushing the
more button.
This is a pain as it's the MOST used button. So having to push two
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