, 9 Jul 2010, Mike Ely wrote:
> (off list)
Continuing to veer off-topic...
> Yes indeed we do. The telcos here are absolutely abhorrent, to the
> point that much could be written about how horrible they are but nobody
> would want to read such depressing material. And consumer
On 7/9/10 3:20 PM, "Gordon Henderson" wrote:
> On Fri, 9 Jul 2010, Mike Ely wrote:
>
>> On 7/9/10 9:57 AM, "Mike Ely" wrote:
>>
>>> Hello, list.
>>>
>>> I've set up an outbound alerting system to play a recording when sys
PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] False answer() being sent by cellphone
>> providers
>>
>> On Fri, 9 Jul 2010, Mike Ely wrote:
>>
>>
>>>
>>>>
>>>> I
On 7/9/10 10:29 AM, "Mike Ely" wrote:
> Some of the systems blokes might just figure that¹s another collections agent
> and hang up then ;)
>
>
> On 7/9/10 10:09 AM, "Steve Edwards" wrote:
>
>> On Fri, 9 Jul 2010, Mike Ely wrote:
>>
>
Some of the systems blokes might just figure that¹s another collections
agent and hang up then ;)
On 7/9/10 10:09 AM, "Steve Edwards" wrote:
> On Fri, 9 Jul 2010, Mike Ely wrote:
>
>>> >> I've set up an outbound alerting system to play a recording when s
On 7/9/10 9:57 AM, "Mike Ely" wrote:
> Hello, list.
>
> I've set up an outbound alerting system to play a recording when systems go
> down, etc. and I'm noticing that cellphones tend to answer() and then start
> ringing the actual handset. So far, I've
Hello, list.
I've set up an outbound alerting system to play a recording when systems go
down, etc. and I'm noticing that cellphones tend to answer() and then start
ringing the actual handset. So far, I've verified this behavior with
Verizon, T-Mobile, and Google Voice (the last produces a SERIOU
On 7/8/10 5:07 AM, "Paul Belanger" wrote:
> On Wed, Jul 7, 2010 at 2:46 PM, Mike Ely wrote:
>> Maybe I missed something here? SIP users configured within Asterisk can
>> dial out just fine through the trunk. It's just when I try to use AMI that
>> it fails.
On 7/6/10 8:44 PM, "Mike Ely" wrote:
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com on behalf of Paul Belanger
> Sent: Tue 7/6/2010 5:10 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc:
> Subject:
8:00 PM, Mike Ely wrote:
> Log attached.
>
<--- SIP read from UDP:10.10.10.16:5060 --->
SIP/2.0 401 Unauthorized
> context from sip.conf:
>
> [ShoreTel]
> type=peer
> qualify=yes
> port=5060
> host=10.10.10.16
> context=incoming
> canreinvite=no
>
Your
port=5060
host=10.10.10.16
context=incoming
canreinvite=no
On 7/6/10 4:21 PM, "Paul Belanger" wrote:
> On Tue, Jul 6, 2010 at 4:10 PM, Mike Ely wrote:
>> Obviously, I'm playing around with the context a bit but for now just want
>> to get the outbound call work
SIP user => Asterisk 1.6 server => SIP Trunk => external destination:
works
AMI script => Asterisk 1.6 server => SIP Trunk => external destination:
Failed to authenticate on INVITE to '"asterisk"
;tag=alphanumeric'
I¹ve tried doing things like ³include => contextwithtrunk" in various
places, goog
aside the fact that I don't use Outlook, Ubuntu 8.04 isn't
exactly going to be a security improvement over what I already use.
On 7/1/10 1:19 PM, "Mike Ely" wrote:
> As an interesting aside, every email I get on this list coming from Tilghman
> Lesher is marked with
As an interesting aside, every email I get on this list coming from Tilghman
Lesher is marked with a "To Do" flag by my email client. Every single one.
I don't have any inbound filter that would explain the behavior either.
On 7/1/10 1:15 PM, "Tilghman Lesher" wrote:
> On Thursday 01 July 2010
Yep, I saw that and it's just not the case. I was having it dial my desk
extension, which was decidedly not busy at the time...
On 6/28/10 5:30 PM, "Philipp von Klitzing"
wrote:
>> Well, I¹ve tried this, and something just isn¹t right.
>
> Look here:
>
>> Event: Hangup
>> Channel: SIP/ShoreT
Variable => "Data=/tmp/test.gsm²,
> Exten => 'SIP/170',
> Context => 'accept',
> priority => 1,
> Number
priority => 1,
> Number => 5551212
> Using the accept context, 5551212 is called on DAHDI/1 and user hears
> important.gsm. then they press 1 to hear test.gsm or 2 to hear it later.
>
> Hope this is helpful
>
Hi there,
I¹ve been looking to do an outbound dialer for systems alerting, etc. and
have in large part followed the recipe here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
That and the associated pages at voip-info give a basic set of recipes for
callfiles, but nowhere th
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