Displaying 20 results from an estimated 30000 matches similar to: "dialplan checker"
2011 Oct 18
1
nvfaxdetect in 10.0
Hi gang,
We are moving our 1.4 asterisk with DAHDI over to 10.0 with
SIP. Everything is going nicely except that I can't get NV_FAXDETECT to
compile properly into 10.0. Because of this, I will have to have my
receptionist manually transfer incoming faxes. Any suggestions?
Thanks in Advance
Danny Nicholas
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2009 Jun 01
2
SVN vs "Regular" Asterisk
Hi gang,
Can someone shed some light on the pros and cons of working
with the SVN branches of Asterisk vs working in the 1.4 or 1.6 branches?
What branch does the SVN release roughly equate to?
TIA
Danny Nicholas
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2010 Nov 05
2
Funky IAX behavior between 1.4 and 1.8
Hi Gang,
My production box with my DAHDI cards is a 1.4.26 build. I
have 3 test machines that I do IAX communication with.
Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30.
Machine 2 is a SUSE 11.1 VM running 1.4.30. Machine 3 is another SUSE 11.1
VM running 1.8.0. I can SIP into all 4 machines and life is great. When I
try to IAX from the live machine to
2010 Nov 03
1
Gotoif changed in 1.8?
Hi Gang,
I'm testing 1.8.0 on one of my machines and this snippet
"chokes" on line 7 (works fine with 1.4.30)
[tb-account-balance]
exten => s,1,Set(BALCOUNT=0)
exten => s,n,NoOp(Verbose(acct ${digitacc} pwd ${digitpwd} ))
exten => s,n(runagi),Set(TEST_RETURN="NONE")
exten =>
2010 Sep 14
1
conf checkout
Hi gang,
I see that some posters today don't do full (or any?) backups
of their Asterisk systems/configuration. This may (sort of) help you.
Since pretty much all Linux systems have some sort of PERL installed, these
two files will let you make a quick copy of any configuration or other file
you might be about to change or destroy.
File 1 - /usr/bin/checkout
2009 Jul 20
0
No subject
expected context is valid (may not work on 1.2, I started this ride at 1.4
and therefore have no backward knowledge).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hash Dial
2009 Jan 16
0
No subject
---
span_1 = DAHDI/g1
1,1,dial(${span_1}/${EXTEN:0})
---
I can only presume some form of precedence overrides the group configuration
in the recent asterisk installs and not for the servers set up earlier.
On 26/5/09 4:01 PM, "Kal Feher" <kalman.feher at melbourneit.com.au> wrote:
> Ok I've solved the problem. I do not think it was as switchtype issue after
> all as
2009 Jun 04
2
broken pipe in perl agi
Hi gang,
Since I'm getting no joy from device_Status or SIPPEER in
1.4.26-rc1, I thought I would do an AGI to read my hints and check for line
in use that way. The AGI works fine from a prompt, but returns the dreaded
"utils.c:966 ast_carefulwrite: write() returned error: Broken pipe" when I
try to run it from the dialplan. Here is my dialplan snippet;
2010 Sep 08
0
Requirement or just Best Practice
Hi gang,
I'm in the process of documenting an 1100 line IVR dialplan. In
this dialplan I have almost 300 Set commands. The numeric ones are simple;
Exten => s,1,Set(TYPE3=0)
My question is about the Alpha/Text values;
If I do
Exten => s,1,Set(beep=beep)
Or
Exten => s,1,Set(beep="beep")
Then
Exten => s,n,playback(${beep})
It seems that I get the
2009 Jul 20
2
What am I doing wrong?
Hi Gang,
I've got the latest SVN branch of 1.4 downloaded onto SUSE
11.0. Everything is happy EXCEPT, I can't get fax to be recognized by make
menuselect. I tried copying app_rxfax.c and app_txfax.c to the apps
directory and starting again from ./configure, but no joy. Any suggestions?
Danny Nicholas
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2010 Apr 28
0
command-line dialplan "compiler"
Hello listers,
Still plodding along in the 1.4 tree, though I've started
to dabble in 1.6 land. Today's adventure involves a 2600 line dialplan. My
friend Google only points me to an antique java script and a bunch of GUI
dialplan creators. What is out there that will point out dialplan
errors/problems besides just watching the CLI output and trying to figure
out with
2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your =
recipient is using a codec that isn't ulaw or alaw).
=20
_____ =20
From: asterisk-users-bounces at lists.digium.com =
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel =
freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
2009 Jul 06
1
Bug or Not?
Hi gang,
When I try to park a call using blind-transfer (#1), the
caller hears the lot instead of the transferring party. Attended transfer
and blind transfer from the phone buttons (Polycom 501) work fine, so this
isn't a showstopper, just a "WHY??". Thanks for you attention.
Danny Nicholas
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2009 Jul 20
0
No subject
might be your best bet to get the information you want. I'd look at
voip-info.org for information.
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, September 16, 2009 9:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to list ongoing calls
2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
2009 Jan 16
0
No subject
adding gsm or just comment out the disallow and the 2 allows. (your
recipient is using a codec that isn't ulaw or alaw).
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, January 28, 2009 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
2011 Sep 19
1
oddity with CISCO CCM and Asterisk
Hi List,
I have a system that connects into Asterisk 1.4.41 using CISCO
CCM 7. Everything works great except when a call is transferred to the
operator. The call goes to the operator via a native bridge and is
completed, then a "phantom process" starts and tries to launch a new call
every 15 minutes. I modified the dialplan to hangup these phantom calls,
but no still
2010 Dec 02
4
DAHDI on VMWARE
Hi gang,
We are moving our computers from a cluster of physical machines
to a VMWARE server with virtual machines. We investigated and are looking
to replace our TDM400P/TDM410P with AEX410P cards. Can we run asterisk with
the DAHDI drivers from one of the Virtual machines or is DAHDI going to have
to be a native process on the "REAL" machine?
Thanks
Danny Nicholas