I'm trying to set a CDR userfield to a custom value. This value may contain a
'|' but it's really just part of the value.
However, Asterisk keeps warning me about the application delimiter not being a
pipe.
It's NOT an application delimiter (it's just part of a variable value) so I'm
expecting
Hello,
I noticed that when i move a call file to outgoing directory, two asterisk
threads are dealing with it.
]# grep FAX_44731.call /var/log/asterisk/full.2
[Nov 27 09:23:10] WARNING[26842] pbx_spool.c: Unable to set utime on
/var/spool/asterisk/outgoing/FAX_44731.call: Operation not
There's no priority in your call file.
Sent from my iPhone
On 29/11/2012, at 11:12 PM, Necati Demir nde...@demir.web.tr wrote:
Hello,
I noticed that when i move a call file to outgoing directory, two asterisk
threads are dealing with it.
]# grep FAX_44731.call /var/log/asterisk/full.2
Should I use priority in call files? How the lack of priority causes this
problem?
On 29 November 2012 12:48, Matt Riddell (lists) li...@venturevoip.comwrote:
There's no priority in your call file.
Sent from my iPhone
On 29/11/2012, at 11:12 PM, Necati Demir nde...@demir.web.tr wrote:
Priority is a required parameter. In your call file you are telling Asterisk
to
Channel: DAHDI/g0/0312xxx
MaxRetries: 0
RetryTime: 60
Context: asteriskgw_fax
Extension: s
Go to context asteriskgw_fax, extension s. Priority tells Asterisk where to
start in asteriskgw_fax. Since C
Thanks, i will add priority and see the results.
On 29 November 2012 17:00, Danny Nicholas da...@debsinc.com wrote:
Priority is a required parameter. In your call file you are telling
Asterisk to
Channel: DAHDI/g0/0312xxx
MaxRetries: 0
RetryTime: 60
Context: asteriskgw_fax
On Nov 29, 2012, at 3:54 AM, Vieri wrote:
I'm trying to set a CDR userfield to a custom value. This value may contain a
'|' but it's really just part of the value.
However, Asterisk keeps warning me about the application delimiter not being
a pipe.
It's NOT an application delimiter (it's
Hello
I have been reading the sample extension.conf
;###
[outbound-freenum2]
; This is the handler which performs the dialing logic. It is called
; from the [outbound-freenum] context
;
exten = _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same =
As I understand it, same = is a way to shorthand your list of the other
keywords. In the example you posted, you save 4 keystrokes for each line you
enter; not a lot of savings for this short example, but put it in a 1000+
line dialplan and it's quite a time-saver.
From:
Shitian Long wrote 29.11.2012 18:40:
There is a part of dial
plan from sample extension.conf above. My Question is how same = key
word works .
Thanks
same is used for complex templates, if you
don't want to copy previous line or afraid you can make a typo.
exten
= _1XXNXXX,1,Answer
Hello,
For an operator, I'm looking for a software application with which operator
would be both able:
- to see the list of awaiting calls,
- to fill a (customizable) form with the name, number and reasonto use
whern returning the call.
Suggestions ?
Regards
--
On 29/11/2012 11:47 AM, Salman Zafar wrote:
It is self explanatory, for example:
exten = _X.,1, Noop(Let say we have allowed all numbers i.e. _X
means and . specifies any range)
same = n,NoOp(Here we have skipped mentioning dial-pattern again and
thats it)
Hope I have answered your
That is a good answer.
Thanks.
Any reason why it is not documented?
Ron
On 29/11/2012 11:52 AM, Mikhail Lischuk wrote:
Shitian Long wrote 29.11.2012 18:40:
There is a part of dial plan from sample extension.conf above. My
Question is how same = key word works .
Thanks
same is used
The Wiki is (always) out of date. You might consider taking a look at
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#DialplanBasics_id262049
which is likely less out of data.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On Nov 29, 2012, at 11:18 AM, Ron Wheeler wrote:
That is a good answer.
Thanks.
Any reason why it is not documented?
It's documented on the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Contexts,+Extensions,+and+Priorities
Ron
--
David M. Lee
Digium, Inc. | Software
FOP (Flash Operator Panel) can probably do this for you. Personally I would
do this with Perl, but other posters prefer C or PHP for this type of roll
your own function.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent:
Excellent.
It appears that Getting Started has a lot more stuff in it than the
documentation for 1.8.
Very helpful.
Ron
On 29/11/2012 12:31 PM, David M. Lee wrote:
On Nov 29, 2012, at 11:18 AM, Ron Wheeler wrote:
That is a good answer.
Thanks.
Any reason why it is not documented?
It's
Hi,
I'd like to replace my current VOIP provider with an Asterisk based
solution. I have some ideas I want to run by the list to see if they
are possible, and get answers to a couple questions.
I want to setup two Asterisk servers that are linked to each other:
- The first server would be
On 29/11/12 6:33 pm, Dyweni - Asterisk-Users wrote:
I want to setup two Asterisk servers that are linked to each other:
- The first server would be my external (public) server and would live
in a real data center. The second server would be my internal
(private) server and would live in my
On Friday, November 30th, 2012, the Asterisk community services listed below
will be undergoing maintenance (migration to a new server and software
upgrades). The services will be shut down at approximately 10:30 AM CST (4:30
PM December 1st UTC), and should return no later than 11:30 AM CST.
On Friday, November 30th, 2012, the Asterisk community services listed below
will be undergoing maintenance (migration to a new server and software
upgrades). The services will be shut down at approximately 10:30 AM CST (4:30
PM December 1st UTC), and should return no later than 11:30 AM CST.
Hello List,
Since I'm looking for a new VoIP provider for US origination/termination, I
will very appreciate if you can chare your experience with Flowroute,
Vitelity and Voip.ms
Thanks in advance!
Elder D. Arohuanca
dCAP 1497
Lima - Peru
--
On Thu, Nov 29, 2012 at 3:22 PM, Daniel - Asterisk earohua...@gmail.comwrote:
Hello List,
Since I'm looking for a new VoIP provider for US origination/termination,
I will very appreciate if you can chare your experience with Flowroute,
Vitelity and Voip.ms
Vitelity is reliable and decent,
Several in our group use voip.ms and have no complaints at all
We had a few hickups when Sandy rolled through NYC ( we are all on the NYC
server ) but voip.ms responded quickly to mirror to Seattle and there was
little downtime, and what was lasted a very short time on one day.
voip.ms was
At 02:22 PM 11/29/2012, you wrote:
Since I'm looking for a new VoIP provider for US
origination/termination, I will very appreciate if you can chare
your experience with Flowroute, Vitelity and Voip.ms
I started using Flowroute in Jan 2009 and have been very happy with
their service. I'm
From: Jody Gugelhupf knuef...@yahoo.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Sent: Thursday, November 29, 2012 10:23:01 PM
Subject: audio trouble with asterisk, help very much appreciated
Hi there :)
first about my setup,
hello:I used libpri 1.4.12 version with asterisk 1.8.7, after the pcap files, i
found thatin wireshark setup message, the number type always changed from
subscriber to national number.i have set pridialplan= local and
prilocaldialplan=local in chan_dahdi.conf already. because that, the
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