Zaheer,
On 28/11/07 9:28 AM, "Zaheer K. Master" <[EMAIL PROTECTED]> wrote:
> Yes I have a sip.conf, contents as follows:
>From the CLI can you confirm SIP is running by pasting the results of
'module show like sip'
Cheers
Nick.
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--Bandwidth and
Morning All,
Has anyone here successfully implemented skills based routing within queues?
The concept behind skills based routing is fairly straight forward, and I
know I could do it with multiple queues, agent penalties and a bit of AGI to
put the call into the right queue.
However doing this i
Yes, that will work fine Zaheer.
On 16/10/07 1:32 AM, "Zaheer Master" <[EMAIL PROTECTED]> wrote:
> Hi all,
> If I have 2 single-line SIP phones, I can still do a conference call using
> Asterisk, right? For example, two people in my office are on the call, along
> with 1 other person at a remote
Morning All,
Just wondering if anyone can confirm that peridoic-announce and
periodic-announce-frequency are still valid options within queues.conf?
For testing purposes my queue includes;
periodic-announce-frequency = 10
periodic-announce = demo-congrats
When in the queue however I'm not heari
Scrap that. I've somehow broken all queue announcements including position
and holdtime.
Will repost when I sort out what I've done.
On 24/10/07 11:13 AM, "Nick Brown" <[EMAIL PROTECTED]> wrote:
> Morning All,
>
> Just wondering if anyone can confirm th
Morning All,
Quick question that has me stumped. Have a queue with several members
(Statically defined in queues.conf at this stage for testing) who use Cisco
7960's.
The queue is configured to use rrmemory and generally this works correctly.
However if a member is already on a call their phone w
11:57 AM, "Eric Merkel" <[EMAIL PROTECTED]> wrote:
> On 11/4/07, Nick Brown <[EMAIL PROTECTED]> wrote:
>> Morning All,
>>
>> Quick question that has me stumped. Have a queue with several members
>> (Statically defined in queues.conf at this stage
Another quick question (Spending the day trying to get this project sorted
and tucked away) If I am dynamically adding queue members, they will not
abide to settings within agents.conf will they?
Ie. I need the equivalent of Autologoff however want my agents to receive
calls when someone joins the
Afternoon All,
Today rolled a pre-production box from Trunk back to 1.4.7 (In an attempt to
get a working SCCP channel). During the process Music On Hold appears to
have died (Not, just when calling from a SCCP device, but coming in on SIP
also).
CLI is showing
-- Executing [EMAIL PROTECTED]
t; PaulH
>
>
> On Tue, 2007-11-13 at 15:04 +1100, Nick Brown wrote:
>> Afternoon All,
>>
>> Today rolled a pre-production box from Trunk back to 1.4.7 (In an
>> attempt to get a working SCCP channel). During the process Music On
>> Hold appears to have d
"Paul Hales" wrote:
> Is it possibly a funny zaptel issue? Paul Hales AsteriskIT > > On Tue,
2007-11-13 at
> 15:04 +1100, Nick Brown wrote: > >> Afternoon All, > >> > >> Today rolled a
> pre-production box from Trunk back to 1.4.7 (In an > >
elephone
> (407) 839-0120 - Main Office
> (407) 841-9726 Fax
> http://www.rissman.com/ <http://www.rissman.com/>
>
>
>
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gt; Is there a standard way to say "hid my number"?
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>
> Mike
>
>
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Afternoon All,
Is anyone aware of a way to generate ringing as opposed to starting music on
hold for the party originating a call with followme?
I'm assuming its doable as it looks like FreePBX users get the option (Not to
say that FreePBX haven't got their own followme implementation though).
Is there a need to do it within the dialplan? If not you will find it easier to
do it within AGI. Either connecting directly to the DB or in our case our
developer build a web service which I make SOAP calls to.
Nick.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...
Morning All,
My Google skills may be failing me as I can see people asking this but no
useful responses, I need a way to prioritise calls across queues - I can think
of ways to do this but they are far from elegant and this seems like such a
simple request I am sure I am missing something obvio
Hi All,
Facing an issue at the moment with setting the TOS on packets - the
documentation is a bit light, however is straightforward so unsure if this is a
configuration issue or a bug.
Following is set in sip.conf;
tos_sip=CS3
tos_audio=EF
And is reflected in the CLI;
IP ToS SIP:
Blocking SIP traffic is still going to break ENUM.
The problem with your suggestion Norbert is that Asterisk still would have to
process the requests at an application layer, providing no real advantage to
users of boxes with no grunt.
You could potentially write something to do inspection on
Do you see the issue when calling between two softphones? Do you see the issue
if you call from your mobile into an echo test?
Setting TOS flags on packets will make no difference unless the gear in between
is configured to treat them differently. Not that I envision this is the issue
at all.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Tuesday, 3 August 2010 1:58 PM
To: j...@sunfone.com; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] What do you use for Invoicing?
M
Depends what its connected to
Nick.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Thursday, 12 August 2010 9:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
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