On Tue, Aug 31, 2010 at 9:34 AM, Danny Nicholas da...@debsinc.com wrote:
Why not just copy the _1NXXNXX line into the remote context?
Well, that could be done, and probably would be a good tactic if you have
lots of DID's
and want to do db lookup or something to direct the next call leg.
Here's the updated debug log.
http:/www.computerworkx.net/client/Document.txt
On 8/30/2010 2:55 PM, Paul Belanger wrote:
On Mon, Aug 30, 2010 at 2:48 PM, Todd Reesetrees...@gmail.com wrote:
Thanks for pointing out the misspelling. I've corrected that and still no
luck.
Create a new
On Tue, Aug 31, 2010 at 9:16 AM, Todd Reese trees...@gmail.com wrote:
Here's the updated debug log.
[Aug 30 15:27:41] NOTICE[11568] chan_sip.c: Call from '150' to
extension '6789542133' rejected because extension not found in context
'remote'.
So, again, a context problem. You can confirm by
asterisk*CLI dialplan show 6789542...@remote
There is no existence of 'remote' context
Command 'dialplan show 6789542...@remote' failed.
asterisk*CLI
On 8/31/2010 9:58 AM, Paul Belanger wrote:
dialplan show 6789542...@remote
--
From extensions.conf
[remote]
include = from-internal
include = dialout1
include = dialout2
include = dialout3
include = intercom
exten = 150,1,Macro(oneline,${EXTERNPHONE0})
[dialout1]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten =
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Reese
Subject: Re: [asterisk-users] help with dialplan
asterisk*CLI dialplan show 6789542...@remote
There is no existence of 'remote' context
Command
Todd--
There is probably some nifty anti-infinite-recursion code in the
extensions.conf parser,
to keep asterisk from going into infinite loops trying to descend into the
right context.
In your dialplan, [remote] includes dialout1, dialout2, dialout3, and each
of those
include remote.
I had already check on this. Thanks for the info, though.
On 8/31/2010 10:36 AM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Reese
Subject: Re: [asterisk-users] help
Interesting things going on herel.
After your suggestions, Steve. I reran the dialplan show
16789542...@remote command with the below results.
Phone calls are geting the 404 error and the NOTICE on the console.
[Aug 31 11:06:46] NOTICE[11884]: chan_sip.c:20161 handle_request_invite:
Call
On Tue, Aug 31, 2010 at 9:14 AM, Todd Reese trees...@gmail.com wrote:
Interesting things going on herel.
After your suggestions, Steve. I reran the dialplan show
16789542...@remote command with the below results.
Phone calls are geting the 404 error and the NOTICE on the console.
[Aug 31
Why not just copy the _1NXXNXX line into the remote context?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the
On Mon, Aug 30, 2010 at 10:14 AM, Todd Reese trees...@gmail.com wrote:
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
Post a debug log of the problem:
Todd
How do you have the context in the phones sip configs set?
Bryant
From: Todd Reese trees...@gmail.com
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on
Here is the sip.conf portion for extension 150
[150]
deny=0.0.0.0/0.0.0.0
type=friend
secret=1234567890
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/150
context=from-trunk
canreinvite=no
callgroup=
callerid=device 150
accountcode=
Here's a debug for extension 150
[Aug 30 11:34:53] VERBOSE[2099] config.c: == Parsing
'/etc/asterisk/logger.conf': [Aug 30 11:34:53] DEBUG[2099] config.c:
Parsing /etc/asterisk/logger.conf
[Aug 30 11:34:53] VERBOSE[2099] config.c: == Found
[Aug 30 11:34:53] VERBOSE[2099] logger.c:
: Todd Reese trees...@gmail.com
Sent: Monday, August 30, 2010 11:20 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] help with dialplan
Here is the sip.conf portion for extension 150
[150]
deny=0.0.0.0/0.0.0.0
type=friend
secret=1234567890
qualify=yes
port=5060
pickupgroup=
permit
the desired results.
Bryant
*From*: Todd Reese trees...@gmail.com
*Sent*: Monday, August 30, 2010 11:20 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] help with dialplan
Here is the sip.conf portion
On Mon, Aug 30, 2010 at 11:42 AM, Todd Reese trees...@gmail.com wrote:
Here's a debug for extension 150
In the future, simply attach your debug log to your email. Here is
your problem:
[Aug 30 11:34:55] NOTICE[2079] chan_sip.c: Call from '150' to extension
'6789542133' rejected because
possibly check you spelling: [from-interal] - [dialout1]
include = from-internal
??
On Mon, Aug 30, 2010 at 10:14 AM, Todd Reese trees...@gmail.com wrote:
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My
I actually found that one and corrected it. I have replaced the
context with the from-internal, remote, and dialout1. Each has produced
the same results of a 404 error.
On 8/30/2010 2:10 PM, Paul Belanger wrote:
On Mon, Aug 30, 2010 at 11:42 AM, Todd Reesetrees...@gmail.com wrote:
Thanks for pointing out the misspelling. I've corrected that and
still no luck.
On 8/30/2010 2:33 PM, Alex Bell wrote:
possibly check you spelling: [from-interal] - [dialout1]
include = from-internal
??
On Mon, Aug 30, 2010 at 10:14 AM, Todd Reese trees...@gmail.com
On Mon, Aug 30, 2010 at 2:48 PM, Todd Reese trees...@gmail.com wrote:
Thanks for pointing out the misspelling. I've corrected that and still no
luck.
Create a new debug log with your recent changes, re-attach it the list.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
I have a small system, server, client and 2 phones. Phones are polycom
501's.
In general all is working fine. I can call the two phones, speak etc...
I can have the server call each phone and play a wave file.
However, when trying to setup a direct dial number of 1044 that
calls another machine
Jerry Geis wrote:
I have a small system, server, client and 2 phones. Phones are polycom
501's.
In general all is working fine. I can call the two phones, speak etc...
I can have the server call each phone and play a wave file.
However, when trying to setup a direct dial number of 1044 that
Are your polycom phones set up for overlap dialing or do you dial the
number then press a key to dial?
From you message I tried a couple things...
Clicking New call, then starting to dial this is when it messes up.
when I start entering the number first then click dial this successfull
Jerry Geis wrote:
How do I turn off this overlap dial?
You need to review the dialing rules for the Polycoms.
They'd be located in the ftp directory that you've setup for your
Polycoms to pull their configs from. It's located in the sip.cfg.
Look for the line:
digitmap
On Fri, Nov 7, 2008 at 11:20 AM, Brent Davidson [EMAIL PROTECTED]
wrote:
Jerry Geis wrote:
Are your polycom phones set up for overlap dialing or do you dial the
number then press a key to dial?
From you message I tried a couple things...
Clicking New call, then starting to dial
On Fri, Nov 7, 2008 at 11:27 AM, Doug Lytle [EMAIL PROTECTED] wrote:
Jerry Geis wrote:
How do I turn off this overlap dial?
You need to review the dialing rules for the Polycoms.
They'd be located in the ftp directory that you've setup for your
Polycoms to pull their configs from. It's
Steve Totaro wrote:
For two phones, I would just use the web interface.. That is of
course if you plan on keeping a small amount of phones.
Or, if you absolutely hate the web interface :-P
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
Jerry Geis wrote:
Are your polycom phones set up for overlap dialing or do you dial the
number then press a key to dial?
From you message I tried a couple things...
Clicking New call, then starting to dial this is when it messes up.
when I start entering the number first then click
Thanks Alex, it looks like you had a great answer to the issue at hand. On 10/17/06, Alex Robar [EMAIL PROTECTED]
wrote:If the order is giving you problems, create two separate outbound routes, one for local calls and one for long distance. Make sure the local route is before the LD route, and it
This was posted at The Asterisk Blog ForumsClick here for the original post.
I need someone to explain how the dialplan rules
work? I'm having a hard time getting it. I know that to dial out you
need a 9 and to ignore that 9 once your out... requires a switch of
sorts that tells asterisk to ignore
If the order is giving you problems, create two separate outbound routes, one for local calls and one for long distance. Make sure the local route is before the LD route, and it should work for you. Both outbound routes can use the same trunk without issue.
AlexOn 10/17/06, Chris Ramsey [EMAIL
Cosmin Prund wrote:
I've got a Mobile-to-PBX gateway installed and I want the ability to dial
from my mobile phone into my PBX and next dial a land-line from the PBX so I
can make cheep mobile-to-land-line calls while on the go.
I've contemplated using the WaitExten application but it only
I've got a Mobile-to-PBX gateway installed and I want the ability to dial
from my mobile phone into my PBX and next dial a land-line from the PBX so I
can make cheep mobile-to-land-line calls while on the go.
I've contemplated using the WaitExten application but it only seems to wait
for ONE
On Tue, Nov 08, 2005 at 07:38:20AM -0500, Frank Tarczynski exclaimed:
Since this is my DID, I want the line to ring as normal but allow a user
to breakout and ultimately get an outgoing line.
In this way the DID line would function as a normal telephone line. A
point lost on many responders!
,Congestion
Message: 21 Date: Mon, 7 Nov 2005 14:25:50 -0500 (EST) From: Frank
Tarczynski [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Help
with dialplan to allow breakout to DISA To:
asterisk-users@lists.digium.com Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain;charset=iso-8859-1 Yes
I'm trying to set-up a dialplan for incoming calls that allows a breakout
by pressing something like *. Users would then be able to get an inside
dial tone for voicemail, outgoing calls, etc.
I've been struggling with Waitexten(), Disa() in the dialplan but not
having much luck.
Are there any
I do it this way:
exten = *, 1, Authenticate(PASSWORD)
exten = *, 2, DISA(no-password|DESTINATION_CONTEXT)
exten = *, 3, Hangup
It seems to work fine...
-Rusty
On 11/7/05, Frank Tarczynski [EMAIL PROTECTED] wrote:
I'm trying to set-up a dialplan for incoming calls that allows a breakoutby
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