There are multiple ways to do time-of-day routing.
ExecIf w/ IFTIME, GotoIfTime, and ExecIfTime.
I put some examples below.
Sincerely,
Brian LaVallee
On 9/12/14, 10:05, Eric Wieling wrote:
See ExecIf in the output of "core show applications". The IF function might be useful,
see
This would NOT load)
; -- The parser stopped loading anything past the above mistake --
; -- Missing that space started a block-comment - Arghhh! --
exten => _4X.,1,NoOp(This would NOT load either)
; -end
Guess I have to change my highlight syntax, avoiding dashes in the future.
Sin
On 8/11/14, 11:31, Matthew Jordan wrote:
On Sun, Aug 10, 2014 at 5:02 PM, Deepak Rawat
wrote:
On Mon, Aug 11, 2014 at 3:29 AM, Deepak Rawat
wrote:
Hi,
I modified the query in res/res_config_odbc.c.
Original: "SELECT MAX(LENGTH(var_val)) FROM %s WHERE filename='%s'"
Modified: "SELECT MAX(
basic features
(hold, transfer, redial) are available by default. To duplicate the
digital PBX features you're looking for, will involve two groups of
settings. Configuration on the server -and- configuration on the phone.
SIP phones are NOT dumb terminals, you ha
ITE,ACK,CANCEL,BYE,REGISTER,PRACK,UPDATE
Accept: application/sdp
Sincerely,
Brian LaVallee
On 6/25/14, 11:30 PM, Rafael Visser wrote:
> Hi gurus!!!
>
> I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn
> Every minute asterisk sends an OPTION Request, i beleived
looking to manipulate the ISDN message via SIP, it all
comes down to how the gateway handles the desired functions.
Sincerely,
Brian LaVallee
On 6/26/14, 11:24 PM, Positively Optimistic wrote:
> We're using a Earthlink PRI converted to SIP via a MediaGateway. I assume
> the mediagateway wil
Hi Jonas,
While I don't work with queues, but you could playback announce-holdtime
before putting the caller into the queue.
exten => _X.,1,NoOp(Post Queue Announcement)
same => n,Answer()
same => n,Wait(10)
same => n,Playback(announce-holdtime)
same => n,Queue(real_queue)
Brian
On 6/25/
Since there are a number of setting that could be causing the alarm,
AMI/B8ZS, SF/ESF, etc...
Start with a loop-test, make sure the card can communicate with itself
(using the current settings).
Connect the following pins:
01 (RX-) <--> 04 (TX+)
02 (RX+) <--> 05 (TX-)
Sinc
7;ve though about passing the variable between the middle servers in a
SIP message, side communication channel. But, hoping there might be a
simpler solution.
Sincerely,
Brian LaVallee
--
_
-- Bandwidth and Colocation Provided b
it's connecting a sufficient number of PSTN connections to support those
users.
Sincerely,
Brian LaVallee
On 12/18/13, 11:45 PM, bilal ghayyad wrote:
Hello;
Can someone advise me what is the maximum number of users (IP Phones)
that can be supported by asterisk 1.8 or later?
Reg
Are you looking for something like this?
Note: This will continuously go between the two trunks until the caller
hangs up, can be fixed by adding loop counter.
;
; extensions.conf
;
[LOADBALANCE]
exten => _X.,1,NoOp(Connect to least used trunk)
; - show active count
exten => _X.,n,NoOp(Calls:
it works for extensions and does NOT work on the
context field of the sippeers table, is there any field that can be used?
Sincerely,
Brian LaVallee
---===
;# extconfig.conf
;
[settings]
;
sippeers => mysql,database,sippeers
moresippeers => mysql,database,moresippeers
extensions => mys
lly a small unit that handles one or two DS3's.
The advantage comes when you add the 29th DS1. With VT1.5 it's just adding
a single channel, DS3 will require another whole DS3 to get an additional
DS1.
Sincerely,
Brian LaVallee
> From: Nick Khamis
> Reply-To: Asterisk Use
sterisk-users] Initial REGISTER Request: Contains Credentials
> before 401
>
> Brian LaVallee wrote:
>>
>> My SIP provider is not happy that credentials (in the Authorization header
>> field) are provided in the initial REGISTER request.
>>
>> The SIP provide
in the Asterisk community know how to avoid sending the
credentials until AFTER receiving a 401?
Any suggestions would be appreciated!
Sincerely,
Brian LaVallee
# ===
# sip.conf
# Asterisk 1.8.15-cert1
# ---
;
[general]
;
; - trucated
;
regis
It sounds like phpMyAdmin is NOT on the same server as the Asterisk DB.
You will run into a couple possible issues when allowing remote MySQL access
on the Asterisk server,
You will need to set the MySQL user privileges to a specific host or a
wildcard (%).
Most common issue is the firewall, like
Thanks Jeremy!
>
> On 5/9/13 8:21 PM, Brian LaVallee wrote:
>> When qualify is enabled on a trunk, the From line shows "asterisk". See the
>> SIP message below.
>
> I had the same annoyance/issue. fixed it in
> https://issues.asterisk.org/jira/browse/
ld like to keep qualify enabled without sending the other end any
reference to "asterisk".
Can anyone point me to a setting that will change or remove `²asterisk²`
from `FROM:` in the OPTIONS message?
Thanks,
Brian LaVallee
--
/etc/asterisk/sip.conf (Asterisk 1.8.15-cert1)
[general]
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