Re: [asterisk-users] False answer() being sent by cellphone providers

2010-07-09 Thread Mike Ely
, 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

Re: [asterisk-users] False answer() being sent by cellphone providers

2010-07-09 Thread Mike Ely
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

Re: [asterisk-users] False answer() being sent by cellphone providers

2010-07-09 Thread Mike Ely
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&#x

Re: [asterisk-users] False answer() being sent by cellphone providers

2010-07-09 Thread Mike Ely
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: >> >

Re: [asterisk-users] False answer() being sent by cellphone providers

2010-07-09 Thread Mike Ely
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

[asterisk-users] False answer() being sent by cellphone providers

2010-07-09 Thread Mike Ely
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

[asterisk-users]

2010-07-09 Thread Mike Ely
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

Re: [asterisk-users] Can't dial out through AMI

2010-07-08 Thread Mike Ely
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.

Re: [asterisk-users] Can't dial out through AMI

2010-07-07 Thread Mike Ely
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:

Re: [asterisk-users] Can't dial out through AMI

2010-07-06 Thread Mike Ely
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

Re: [asterisk-users] Can't dial out through AMI

2010-07-06 Thread Mike Ely
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

[asterisk-users] Can't dial out through AMI

2010-07-06 Thread Mike Ely
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

Re: [asterisk-users] Remote Party ID issue

2010-07-01 Thread Mike Ely
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

Re: [asterisk-users] Remote Party ID issue

2010-07-01 Thread Mike Ely
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

Re: [asterisk-users] Call file structure and syntax

2010-06-29 Thread Mike Ely
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

Re: [asterisk-users] Call file structure and syntax

2010-06-28 Thread Mike Ely
Variable => "Data=/tmp/test.gsm², > Exten => 'SIP/170', > Context => 'accept', > priority => 1, > Number

Re: [asterisk-users] Call file structure and syntax

2010-06-22 Thread Mike Ely
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 helpfulŠ >

[asterisk-users] Call file structure and syntax

2010-06-22 Thread Mike Ely
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