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]
; - Truncated
[TRUNK
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/ASTERISK-17616
That's looks like the problem I
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,
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
;
register=accountnum
-users@lists.digium.com
Subject: Re: [asterisk-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 provider ONLY wants
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 sym...@gmail.com
Reply-To: Asterisk Users Mailing
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 = mysql,database,extensions
other
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:
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?
Regards
Bilal
. But, hoping there might be a
simpler solution.
Sincerely,
Brian LaVallee
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
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-)
Sincerely,
Brian LaVallee
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 6/25/14,
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 will convert the headers
: INVITE,ACK,CANCEL,BYE,REGISTER,PRACK,UPDATE
snip
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 that it's
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 have to configure them to operate
how you want.
Sincerely,
Brian LaVallee
On 8/11/14, 11:31, Matthew Jordan wrote:
On Sun, Aug 10, 2014 at 5:02 PM, Deepak Rawat
deepaksingh.ra...@gmail.com wrote:
On Mon, Aug 11, 2014 at 3:29 AM, Deepak Rawat deepaksingh.ra...@gmail.com
wrote:
Hi,
I modified the query in res/res_config_odbc.c.
Original: SELECT
)
; -- 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.
Sincerely,
Brian LaVallee
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 core show
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