On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI ad...@tootai.netwrote:
Hi all,
I have to questions about queues. Member is a phone like SIP/myphone and
only one member in the queue.
At first, DIALSTATUS doesn't return any status. How to now if a call in
queue has been answered or if
Hi Satish
Le 03/07/2013 09:15, Satish Barot a écrit :
On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI
ad...@tootai.net mailto:ad...@tootai.net wrote:
Hi all,
I have to questions about queues. Member is a phone like
SIP/myphone and only one member in the queue.
At
Hi all. We have a problem with correct dialstatus and cdr(disposition) when
using call-limit. When call-limit reached dialstatus is CHANUNAVAIL and
CDR(disposition)='NO ANSWER'
-- Executing [0014@sub_pbxdialco:49] Dial(SIP/1295-01f8,
SIP/0014,12,tTkK) in new stack
== Using SIP RTP
On 3 Jul 2013, at 12:28, I.Pavlov wrote:
[2013-07-03 15:22:27] NOTICE[29728]: chan_sip.c:6003 update_call_counter:
Call to peer '0014' rejected due to usage limit of 1
-- Couldn't call 0014
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [0014@sub_pbxdialco:50]
Dear list.
This is probably a complex subject but is that right to consider:
a) each distinct linkedid field value in a mysql CEL table as a unique call?
b) the duration of a call as the period (eventtime fields) between
BRIDGE_END and BRIDGE_START events of the same linkedid sequence? (not
On Tue, Jul 2, 2013 at 11:03 AM, Amit Patkar | ATPL a...@avhan.com wrote:
Hi Matt,
As required, please find DEBUG trace for datetime function. I have used
this function in Dialplan to capture DEBUG trace. I hope, this can help us
in resolving the issue.
[Jul 2 15:54:44] DEBUG[2698]
On Wed, Jul 3, 2013 at 2:37 PM, Administrator TOOTAI ad...@tootai.netwrote:
Hi Satish
Le 03/07/2013 09:15, Satish Barot a écrit :
On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI
ad...@tootai.netmailto:
ad...@tootai.net wrote:
Hi all,
I have to questions about queues.
Thanks for answer. For correct dialstatus I use now:
Set(DIALSTATUS=${IF($[ ${SIPPEER(${EXTEN},curcalls)} =
${SIPPEER(${EXTEN},limit)} ]?BUSY:${DIALSTATUS})});
I tried to use Busy app and got CDR(disposition)=BUSY, but in this way I
can't redirect *calling* channel to voicemail, because it
Le 03/07/2013 15:07, Satish Barot a écrit :
[...]
Then you should add Local channel as a queue member and dial your SIP
member from Local channel context. A little hint here. Suppose you
have a support queue configured in queues.conf
;queues.conf
[support]
... ...
member =
Faced some issues with size. So deleted some content. But this was
full file and we are using default file.
Hi Matt,
I have pasted entire say.conf here. It has datetime extension.
;
; language configuration
;
[general]
mode=new; method for playing numbers and dates
; old - using
Arch = x86_64
OS = CentOS-6.4 (freepbx)
Asterisk = 11.4.0
FreePBX = 2.11.0.2
We use Snom870 handsets with firmware v.8.7.3.19.
I am trying to develop a custom dial plan to invoke a distinctive
ring-tone when an external call is transferred internally. Based on
an earlier solution I discovered I
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I have an Asterisk 11.4 SIP only system. We are using a SIP trunk
for outside calls. We are having a problem with calls dropping after
a transfer.
Outside call awswered by phone 101
101 transfers to 100 (attended transfer)
call is dropped
I tried with hangup cause but my script is not executed... also I tried the
same script with mix monitor itself no sucess.
The script what I have is, am converting wav file to flac format..
On 11 Jun 2013 11:17, Satish Barot satish4aster...@gmail.com wrote:
And yes if you want to use System
On several occassions lately, my home Asterisk box has stopped
registering with my VoIP provider. I haven't been able to reproduce the
problem, and the log doesn't contain anything useful.
How can I increase the log verbosity for SIP registration-related
events? I've looked through logger.conf
I would enable SIP debugging, but only for that provider.
This can be done on the Asterisk command line by using either of the
following:
* sip set debug peer Your VoIP provider peer name
or
* sip set debug ip the ip address of your VoIP provider
On 3 July 2013 23:25, Ian Pilcher
On 07/03/2013 05:51 PM, David Duffett wrote:
I would enable SIP debugging, but only for that provider.
Will that increase the logging verbosity?
This can be done on the Asterisk command line by using either of the
following:
* sip set debug peer Your VoIP provider peer name
or
* sip
See cli.conf.sample
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ian Pilcher
Sent: Wednesday, July 03, 2013 7:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk stops
On Wed, Jul 3, 2013 at 7:40 PM, Administrator TOOTAI ad...@tootai.netwrote:
Le 03/07/2013 15:07, Satish Barot a écrit :
[...]
Then you should add Local channel as a queue member and dial your SIP
member from Local channel context. A little hint here. Suppose you have a
support queue
On Thu, Jul 4, 2013 at 1:30 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
I tried with hangup cause but my script is not executed... also I tried
the same script with mix monitor itself no sucess.
The script what I have is, am converting wav file to flac format..
On 11 Jun 2013
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