On Thu, Jul 4, 2013 at 12:24 AM, James B. Byrne wrote:
> I have this code in a dial plan:
>
> exten => _417XX,n,GotoIf($["${CALLERID(num)}" >
> "SIP/41799"]?notfromlocal)
> exten => _417XX,n,GotoIf($["${CALLERID(num)}" <
> "SIP/41700"]?notfromlocal)
>
> The value of "${CALLERID(num)}" appears to b
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 2
On Wed, Jul 3, 2013 at 7:40 PM, Administrator TOOTAI wrote:
> 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
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 registerin
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
>
> or
>
> *> sip set debug ip
Can
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
or
*> sip set debug ip
On 3 July 2013 23:25, Ian Pilcher wrote:
> On several occassions lately, my home Asterisk box has stopped
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 a
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" wrote:
> And yes if you want to use System application in your dialpl
I have this code in a dial plan:
exten => _417XX,n,GotoIf($["${CALLERID(num)}" >
"SIP/41799"]?notfromlocal)
exten => _417XX,n,GotoIf($["${CALLERID(num)}" <
"SIP/41700"]?notfromlocal)
The value of "${CALLERID(num)}" appears to be "SIP/41712-0181"
-- Executing [41720@from-internal:5] GotoI
-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 af
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
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 ast
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 => Local/1000@member
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, becaus
On Wed, Jul 3, 2013 at 2:37 PM, Administrator TOOTAI wrote:
> 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>> wrote:
>>
>> Hi all,
>>
>> I have to questions about queues. Member is a phone like
On Tue, Jul 2, 2013 at 11:03 AM, Amit Patkar | ATPL 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] chan_sip.c: C
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
cons
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
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 RT
Hi Satish
Le 03/07/2013 09:15, Satish Barot a écrit :
On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI
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 first, DIALSTATUS d
On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI wrote:
> 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 caller jus
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