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can concantenate n tif files into one. You can insert a
page into a specific page number. You can convert from various tiff formats.
Walker
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difference between the TDM100P and the
channel banks listed.
Maybe someone else has some experience to share.
Walker
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.
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to the Dial command and that took care of it.
Walker
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)
or something simlar so as to relinquish the machine but otherwise polling
like crazy?
http://lists.digium.com/pipermail/asterisk-dev/2003-December/002391.html
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by including various contexts that
are required for the IVR.
Walker
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or sip. That's probably why you're seeing v1
verses v6.3 or whatever.
The 7905 and the 7905G both run the 1.01 SIP firmware.
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messages in the /var/log/asterisk/messages file
at the * CLI prompt, enter `logger reload`
Now you will see the dtmf decoded in the debug messages.
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.
Walker
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pointed out, make sure the
device is a `friend`
Walker
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On Mon, Mar 15, 2004 at 09:56:04PM +0100, Olle E. Johansson wrote:
Walker Haddock wrote:
On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote:
The incominglimit limits how many simultaneous calls a UA may place to
Asterisk.
I'm pretty sure that the incominglimit specifies
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Superior **
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On Fri, Jan 30, 2004 at 10:58:15AM -0600, Eric Wieling wrote:
Are you sure this works for VoiceTronix Driver? It's not implemented in
app_dial, but in chan_zap.
I've only used it with a zap device. Sorry I didn't think this through.
On Fri, 2004-01-30 at 10:15, Walker Haddock wrote
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a carrier like MCI, Verizon, etc. The calls placed
would connect anywhere in the world via the USA carrier.
Thanks, Walker
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` command. Are you trying to do it
automatically from a script?
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IAX to connect. I also have an
800 number that dials me.
send an email to [EMAIL PROTECTED]
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1634A
are out of sync. Maybe
left over from the predecesor version.
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-users
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December 2003 23:04, Walker Haddock wrote:
On Tue, Dec 16, 2003 at 07:04:47PM -0600, Brian West wrote:
Its an upppercase W
Actually, it's both. The 'w' is case-insensitive.
OK, still no joy! Here's the way I tried it:
exten = _91NX,1,Dial(ZAP/g1/${EXTEN}WWW555,70)
You
extension.
Walker
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On Wed, Dec 17, 2003 at 12:57:38AM -0600, Steven Critchfield wrote:
On Tue, 2003-12-16 at 23:51, Tilghman Lesher wrote:
On Tuesday 16 December 2003 23:04, Walker Haddock wrote:
On Tue, Dec 16, 2003 at 07:04:47PM -0600, Brian West wrote:
Its an upppercase W
Actually, it's both
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numbers.
So, if I want the internal IAX2 service to listen on port 80 and I want to register
with the external IAX2 server using the standard port, is there another change I need
to make to the IAX2 code?
Thanks
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:
On Tuesday 16 December 2003 18:56, Walker Haddock wrote:
I am working on a project where we must connect to the * server
behind a firewall. They can only open up port 80 udp.
I have managed to get IAX working fine by specifying the port=80 in
the iax.conf file. However I would like
Linux
Thanks, must be something I'm doing wrong!
On Tue, 16 Dec 2003, Walker Haddock wrote:
I have a dialplan that requires that we use * to send the long distance access
code to ATT. I have found in the list that the `w` command can be used to inject
a pause, I have tried
On Wed, Dec 17, 2003 at 12:21:28AM -0500, William Waites wrote:
On Tue, Dec 16, 2003 at 10:59:50PM -0600, Walker Haddock wrote:
edit iax2.h file and change line 73 as follows:
#define IAX_DEFAULT_PORTNO 80/* 4569 */
this is *really* the *wrong* way to fix
off topic, but I had to ask Dan about this specific case, my
appologies.
Walker
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On Mon, Dec 15, 2003 at 05:06:57PM +0200, Dan wrote:
Hi,
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
Dan, you say fax works better on the TDM400 than the ATA186. I'm having
problems with faxing on the TDM400.
I had to drop the rx and tx rates on the fax
If anyone on the list has successfully configured and used the GS
speaker phone,
could use please share
Great fix, replaced with Cisco 7960
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-users
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phone: (216) 533-6967
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On Sun, Dec 07, 2003 at 10:56:38AM -0500, Troy Settle wrote:
-Original Message-
From: Walker Haddock
Sent: Thursday, December 04, 2003 7:54 PM
To: [EMAIL PROTECTED]
We have an installation with 9 inbound voice channels (one is
the fax) and 768K data. It is a Hybrid PRI
On Thu, Dec 04, 2003 at 08:15:21PM -0500, Jim Flagg wrote:
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 04, 2003 7:54 PM
Subject: Re: [Asterisk-Users] Channelbank Recomendation and GS102 question
We have
/
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list
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recently about the details for
getting the sip images and setting them up on the 7960/40 Cisco IP phones.
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/relnotes/31_201.htm
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/asterisk/README.cdr and the /usr/src/asterisk/cdr/cdr_csv.c file. Seems to
work OK.
Walker
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1634A
On Wed, Nov 26, 2003 at 12:12:06PM -0600, Walker Haddock wrote:
On Wed, Nov 26, 2003 at 09:26:18AM -0800, Richard Lyman wrote:
Asterisk wrote:
Hello!
Does anyone know where I can find out about the CDR fields?
I know most of them are self expiatory, but what is disposition
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the installation manual actually referred to.
Walker
We intend to deploy Asterisk in a 72 extensions + 16 trunks in a while,
so any info will be great.
thanks
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? There is an option tftp-server-name, but I can't find an
option for the server IP address.
option tftp-server-name 192.168.254.179;
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zapata.conf put:
group = 1
before the device declaration of Zap/1 and Zap/2.
Walker
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dtmfmode=inband
Walker
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On Mon, Nov 24, 2003 at 03:59:14PM -0500, Andrew Thompson wrote:
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 24, 2003 2:46 PM
Subject: Re: [Asterisk-Users] test call request
Adding canreinvite=no to your sip.conf
.
bart
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://www.cisco.com/en/US/tech/tk652/tk701/tech_protocol_family_home.html
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and flexible.
Walker
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problem.
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On Tue, Nov 18, 2003 at 06:44:23PM -0600, Ling C. Ho wrote:
Is there any way to search the archive? I know this question must has
been asked before but the problem is there don't seems to be a way to
search the archive. Looking through every thread for each month is just
too troublesome.
Basically.
grandstream = 192.168.1.70
asterisk = 192.168.1.1
The error I see is ;-
-- Executing Dial(Zap/2-1, SIP/206|20) in new stack
-- Called 206
-- SIP/206-582e is ringing
NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from
'sip:[EMAIL
; M,D,Y in any order (5 chars max)
keepAlive = 120
I had to put this in to get the voice to go from the 7960 to *:
bindaddr = 192.168.254.179 ; Address to bind to
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characteristics?
Is there any problem with opening them up and replacing the components?
Thanks, Walker Haddock
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for the caller in the campon function. Sample of
extensions.conf
Paul, this looks great. I'd like to try it.
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On Mon, Oct 27, 2003 at 08:12:05PM -0500, Jon Pounder wrote:
look for a T1 failover switch.
(cheap as dirt on ebay, mine was $7.50, - yes really, I got the decimal in
the right spot, hard to find an empty rackmount box that is cheaper.)
What is the manufacturer and model number. I searched
to false
i.e. 0
I expect the expression evaluated to 0 because OFFICEHOURS was not set. So
Set Global to a default value as above.
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more system information,
please let me know.
thanks, Walker
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*/
if (p-owner)
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On Wed, Oct 22, 2003 at 09:46:52AM +1000, Paul Liew wrote:
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 22, 2003 5:50 AM
Subject: Re: [Asterisk-Users] Call Waiting on SIP phones
Paul, I'm getting a patch error when
-4299
No such SIP Call ID 'SIP/206-4299'
I always get the No such SIP Call ID ...
Thanks, Walker
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Linux
I try typing:
sip show channel SIP/[Tab][Tab][Tab]
like with bash and nothing ever pops up.
What I really want to do is figure out what the channel name is so I can use in in the
Channel Status command in the AGI.
regards
Martin
On Tue, 14 Oct 2003, Walker Haddock wrote:
I
0ms ms ULAW
...
8 active SIP channel(s)
asterisk*CLI sip show channel [EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
...
I typed in the Call ID and hit tab, voila!
Thanks, Walker
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday
: (Dial) Options: (SIP/2400SIP/2410)
-- Called 2400
-- Called 2410
-- SIP/2400-3320 is ringing
-- SIP/2410-a8ca is ringing
...
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: (Dial) Options: (SIP/2400SIP/2410)
-- Called 2400
-- Called 2410
-- SIP/2400-3320 is ringing
-- SIP/2410-a8ca is ringing
...
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