Totaro wrote:
Anthony Francis wrote:
Tim Panton wrote:
On 5 Aug 2007, at 06:54, Douglas Garstang wrote:
I don't think creating a network without a single point of
failure
is unreasonable.
It's impossible. I can't think of a single example where
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stephen Bosch
Sent: Tuesday, August 07, 2007 10:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Teliax Quality of Service
Brian Capouch
, 2007, at 10:42 AM, Stephen Bosch wrote:
Eric ManxPower Wieling wrote:
Douglas Garstang wrote:
Let's assume for a moment that it's impossible. That does not
mean adding additional servers and additional networking
equipment does not add value, or is a worthless endeavour.
I agree
:
Anthony Francis wrote:
Tim Panton wrote:
On 5 Aug 2007, at 06:54, Douglas Garstang wrote:
I don't think creating a network without a single point of
failure
is unreasonable.
It's impossible. I can't think of a single example where this
actually exists.
Getting even
Wieling wrote:
Douglas Garstang wrote:
Let's assume for a moment that it's impossible. That does not mean
adding additional servers and additional networking equipment does not
add
value, or is a worthless endeavour.
I agree with that. At least two people that I know run ITSPs. Each
:
Anthony Francis wrote:
Tim Panton wrote:
On 5 Aug 2007, at 06:54, Douglas Garstang wrote:
I don't think creating a network without a single point of
failure
is unreasonable.
It's impossible. I can't think of a single example where this
actually exists.
Getting even
I don't think creating a network without a single point of failure is
unreasonable.
-Original Message-
From: [EMAIL PROTECTED] on behalf of Stephen Bosch
Sent: Sat 8/4/2007 3:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Teliax Quality
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Teliax Quality of Service
On 5 Aug 2007, at 06:54, Douglas Garstang wrote:
I don't think creating a network without a single point of failure
is unreasonable.
It's impossible. I can't think of a single example where
How can I objectively measure jitter in Asterisk on a SIP channel?
I don't just want to turn the new 1.4 jitter buffer on. I want to
measure jitter.
Thanks,
Doug.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
] On Behalf Of Douglas
Garstang
Sent: Friday, August 03, 2007 12:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Measuring Jitter in Asterisk
How can I objectively measure jitter in Asterisk on a SIP channel?
I don't just want to turn the new 1.4 jitter
Aug 2007, Douglas Garstang wrote:
If it COULD, you could leave a tshark process running, constantly
measuring jitter in real time. You'd run one for each ITSP you use,
and
voila, you have real time jitter metrics on a provider by provider
basis.
There are various command-line SIP
-08-03 at 12:31 -0700, Douglas Garstang wrote:
How can I objectively measure jitter in Asterisk on a SIP channel?
I don't just want to turn the new 1.4 jitter buffer on. I want to
measure jitter.
You can use Wireshark (formerly Ethereal) to analyze the RTP stream
after it's been captured
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: Friday, August 03, 2007 2:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
On Fri, 3
-0700 2007/8/3, Douglas Garstang wrote:
How can I objectively measure jitter in Asterisk on a SIP channel?
I don't just want to turn the new 1.4 jitter buffer on. I want to
measure jitter.
Thanks,
Doug.
You could look at the txjitter and rxjitter values (and other values)
stored
-0700 2007/8/3, Douglas Garstang wrote:
How can I objectively measure jitter in Asterisk on a SIP channel?
I don't just want to turn the new 1.4 jitter buffer on. I want to
measure jitter.
Thanks,
Doug.
You could look at the txjitter and rxjitter values (and other values)
stored
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ira
Sent: Thursday, August 02, 2007 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Teliax Quality of Service
At 09:23 AM 8/2/2007,
Don't know about the 320, but we provisioned the 301's. They're
provisioning is basically the same as the 501's and 601's. What problems
are you having?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Doug
Sent: Wednesday, August 01,
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of SIP
Sent: Wednesday, August 01, 2007 1:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Retail DID provider ?
IdeaSIP, Voxbone, Gizmo
: [asterisk-users] Different SIP From and Auth?
Hi
I have asked this questions,but have no answer :) I also want Asterisk
do not check to head with digest username in registration,how can we
do that?
On 7/12/07, Douglas Garstang [EMAIL PROTECTED] wrote:
Is it possible to have Asterisk allow
Is it possible to have Asterisk allow the From address in a SIP invite
to be different to the required digest username?
The auth parameter supposedly allows it, but whether or not I set auth
to be what the UA sends as the digest username, Asterisk just complains
that the from and the digest are
have asked this questions,but have no answer :) I also want Asterisk
do not check to head with digest username in registration,how can we
do that?
On 7/12/07, Douglas Garstang [EMAIL PROTECTED] wrote:
Is it possible to have Asterisk allow the From address in a SIP invite
to be different
I have this in sip.conf:
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
progressinband=yes
[19256002182]
type=friend
username=19256002182
callerid=Test hone 1 +19256002182
host=dynamic
canreinvite=no
secret=password
context=test
I have this in my dialplan...
[general]
static=yes
writeprotect=no
clearglobalvars=no
[start]
exten = 5000,1,Answer
exten = 5000,n,Wait(1)
exten = 5000,n,NoOp(${CALLERID(num)})
exten = 5000,n,Playback(tt-monkeys)
which, when I dial 5000, executes this...
== Parsing
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: Monday, June 18, 2007 5:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 180 Ringing with SDP
On Mon, 18 Jun 2007,
I have this in my dialplan...
[general]
static=yes
writeprotect=no
clearglobalvars=no
[start]
exten = 5000,1,Answer
exten = 5000,n,Wait(1)
exten = 5000,n,NoOp(${CALLERID(num)})
exten = 5000,n,Playback(tt-monkeys)
which, when I dial 5000, executes this...
== Parsing
Can anyone recommend any wholesale SIP termination providers that will
automatically charge a credit card? Most seem to want upfront payment
and a credit balance but that sucks when you have to watch it like a
hawk to make sure the balance never hits zero. I'm looking for a
provider that can
We're dialing a disconnected number via Level 3's vector network, and
are receiving this. The response has SDP in it. Apparently, Level 3 is
playing early media. Asterisk doesn't seem to know what to do with SDP
in a 180 RINGING, and just plays ringing. What am I missing here? How
can Asterisk see
as both
your regcontext and as a context in extensions.conf (or an .ael, or
whatever).
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Douglas Garstang
Sent: Wednesday, June 06, 2007 7:08 PM
To: asterisk-users@lists.digium.com
Subject
I don't know if this is possible, and I can't quite get my head around
how to do it...
If I am using DUNDi for redundancy in a cluster, when Phone1 makes a
call to Phone2, both Asterisk A and B will be in the RTP stream:
+---+ +---+
| A |-| B |
/+---+ +---+\
Does anyone know how the Linksys PAP2T ATA's can be mass provisioned?
Documentation seems to be sketchy, even on the Linksys web site.
Thanks,
Doug.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
How do you get PAP2T-NA's? They aren't even on Linksys's web site.
-Original Message-
From: Doug [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 07, 2007 10:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Douglas Garstang
Subject: Re: [asterisk-users] Provisioning
On Thu, 7 Jun 2007, Douglas Garstang wrote:
Does anyone know how the Linksys PAP2T ATA's can be mass provisioned?
Documentation seems to be sketchy, even on the Linksys web site.
If it's like the pap2, you can use tftp and xml. This should get you
started.
/tftpboot/spa000F66A83C90.xml:
?xml
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jared Smith
Sent: Thursday, June 07, 2007 10:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and reinvites...
On 6/7/07, Douglas
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jared Smith
Sent: Thursday, June 07, 2007 10:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and reinvites...
On 6/7/07, Douglas
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling
Sent: Thursday, June 07, 2007 2:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and reinvites...
Douglas
Ok, I could have sworn this was fixed in Asterisk 1.2, but it seems in
Asterisk 1.4.4, that doing a reload, or even an 'extensions reload' will
clear any extensions that have been created by regexten. This is VERY
bad!
Doug.
___
--Bandwidth and
Where's Steve Murphy when you need him? :-)
This doesn't seem to work in AEL2...
Macro foo(arg1) {
.
Includes {
Hangup;
}
}
The error is: File: /etc/asterisk/extensions.ael, Line 59, Cols: 5-12:
Error: syntax error, unexpected KW_INCLUDES, expecting
I previously worked for a company that did some heavy load testing with
Asterisk on multiple core Sun systems. We saw that no matter how many
cores you threw at Asterisk, it always used ONE core to process calls,
even at very high loads.
-Original Message-
From: [EMAIL PROTECTED]
Speaking of SQLite, is there an Asterisk SQLite command?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Friday, June 01, 2007 9:41 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk mysql support
On Fri,
I'm trying to find a high port count ATA device. We have a lot (up to
110) analog devices that we need to bridge to IP. Rather than go out and
buy 110 ATA's, it would make more sense to buy a single chassis type
device with some number of ports and blades. Anyone know if such a
device exists?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, May 31, 2007 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] High Port Count ATA
I'm trying to find a high port count ATA device. We have
I have a scenario here with IP phones, on a private 192.168 network
connecting to an Asterisk box, also on the same 192.168 private network.
We'd like to have the Asterisk box also be able to send traffic to the
public IP space. For this, we would need to multi-home the box, and put
two network
All,
I am trying to find a SIP ITSP that honors dial around compensation. We
are adding a Flex ANI code to our outgoing SIP invites by appending an
isup-oli tag to our From: address, like this:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
Has anyone tried to compile the current version of MySQLPool from
http://www.yosd.at http://www.yosd.at/ against Asterisk 1.4.4?
It seems to not compile...
[EMAIL PROTECTED] res_mysqlpool]# make
gcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g
-Original Message-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent: Monday, January 15, 2007 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue cmd option 'i'
On 1/15/07, James Fromm [EMAIL PROTECTED] wrote:
Using Asterisk
That's not correct. You need one G729 license for each transcoding instance. If
you have two SIP channels and both are G729, then no license is required. If
you have two SIP channels, and one is G729 and the other is ulaw, then a
license is required.
Doug.
-Original Message-
From:
Does this model give you functioning mwi?
-Original Message-
From: Ray Jackson [mailto:[EMAIL PROTECTED]
Sent: Friday, January 05, 2007 3:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voicemail personalised greetings using
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Core Dump in app_queue - Anyone
seen?
Douglas Garstang wrote
*snipped
cat = 0x81507e0 mcao_QMain
tmp = 0x6d6f7250 Address 0x6d6f7250 out of bounds
*snipped
a quick run through
Anyone seen this? It ocurred on a 'reload app_queue.so' command.
Asterisk version is 1.2.9.1.
Tried again, but it was not immediately reproducable.
Doug.
(gdb) bt
#0 reload_queues () at app_queue.c:3339
#1 0xb778a7a8 in reload () at app_queue.c:4012
#2 0x0805bb44 in ast_module_reload
Bugger. :(
-Original Message-
From: Douglas Garstang
Sent: Wed 1/3/2007 3:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject:[asterisk-users] Asterisk Core Dump in app_queue - Anyone seen?
Anyone seen this? It ocurred on a 'reload
Can someone tell me how Asterisk handles music-on-hold between servers?
Documentation for this is non-existent.
Lets say user A, who is registered on pbx1, calls user B, who is registered on
pbx2.
1. User A puts user B on hold. The moh that is played to user B should be
specified according to
I don't think that's possible. We have the same issue.
-Original Message-
From: Dovid B [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 27, 2006 8:04 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom 601 Contacts List
Good morning,
I have a Polycom 601 with
You could put together a web page that talks to the Asterisk Manager.
-Original Message-
From: Rob Hillis [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 26, 2006 11:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Agent presence
Hi
SIP presence to the degree required.
Douglas Garstang wrote:
You could put together a web page that talks to the
Asterisk Manager.
-Original Message-
From: Rob Hillis [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 26, 2006 11:48 PM
To: Asterisk Users Mailing List
it will obey. I'll
try and find it later if you haven't found it by the time I get to the office.
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang
Sent: Wednesday, December 27, 2006 10:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
You can only search a month at a time... :(
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 27, 2006 10:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Searching the list
Mark Greene wrote:
Sounds great. What's the mechanism by which Asterisk servers communicate the
mwi status between them?
-Original Message-
From: Jean-Yves Avenard [mailto:[EMAIL PROTECTED]
Sent: Monday, December 25, 2006 11:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Well, this is weird.
After receiving a sip subscribe message from peer 2529266, here's what Asterisk
responds with:
-- (14 headers 0 lines)---
Found user '2529266'
Looking for 2943110 in bell_CallStart (domain ua2.ipt.xxx.com)
Dec 26 10:19:34 NOTICE[27345]: pbx.c:1741 pbx_extension_helper:
Well there's ya problem.
If 2943110 doesn't have a match in the dialplan anywhere, Asterisk pukes.
What's up with that? I don't see why that is necessary.
Doug.
-Original Message-
From: Douglas Garstang
Sent: Tuesday, December 26, 2006 10:25 AM
To: Asterisk Users Mailing List - Non
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP Subscription Bug?
Douglas Garstang wrote:
Well there's ya problem.
If 2943110 doesn't have a match in the dialplan anywhere, Asterisk
pukes. What's up with that? I don't see why that is necessary
Subscription Bug?
On 26/12/06, Douglas Garstang [EMAIL PROTECTED] wrote:
To put it generically, if user A subscribes to the status
of user B, and there is no dialplan match for user B, then
Asterisk will return 404 Not Found to user A.
Yes, because the subscribe is against an extension, which
Don't bother. If the version of asterisk the crash ocurred in isn't the latest,
the moderators will close the bug.
-Original Message-
From: Vicky [mailto:[EMAIL PROTECTED]
Sent: Friday, December 22, 2006 6:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I'm no C programmer, but is this 32 limit just an array definition somewhere?
Wouldn't it be a no brainer to track it down and increase it so some very large
number?
-Original Message-
From: John Harragin [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 21, 2006 11:56 AM
To:
-Original Message-
From: Richard Lyman [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Match a Numer - then continue with,
dialplan
Douglas Garstang wrote
-0700, Douglas Garstang wrote:
-Original Message-
From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 4:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Match a Numer - then continue with
dialplan
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 6:16 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG So
Yes, we have issues with this application being removed as well. In my opinion,
it's a loss of functionality.
-Original Message-
From: Markus Bönke [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 6:40 AM
To: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]
-Original Message-
From: Doug Crompton [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 8:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: Match a Numer - then continue with
dialplan
I haven't really been following
-Original Message-
From: Andreas Sikkema [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Match a Numer - then continue with
dialplan
[snip]
[coo1_CallStart]
-Original Message-
From: Gavin Hamill [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 7:10 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] AgentCallbackLogin() deprecated in 1.4
On Wed, 20 Dec 2006 14:39:42 +0100
Markus Bönke [EMAIL PROTECTED]
-Original Message-
From: Andreas Sikkema [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 9:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Match a Numer - then continue with
dialplan
Bzzt. In order to call SetVar, I
-Original Message-
From: Peter Bowyer [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 9:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Match a Numer - then continue with
dialplan
On 20/12/06, Douglas Garstang [EMAIL
-Original Message-
From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Match a Numer - then continue with
dialplan
Douglas Garstang wrote
-Original Message-
From: Douglas Garstang
Sent: Wednesday, December 20, 2006 10:54 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Match a Numer - then continue with
dialplan
-Original Message-
From: Eric ManxPower
-Original Message-
From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 10:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Match a Numer - then continue with
dialplan
Douglas Garstang wrote
-Original Message-
From: David Gomillion [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 10:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with,
dialplan
I think you're making it far too difficult.
What I do
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 11:47 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
In article
[EMAIL PROTECTED],
Douglas Garstang [EMAIL
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 11:47 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
In article
[EMAIL PROTECTED],
Douglas Garstang [EMAIL
What about comparing the caller id to the dialled number, and if they match,
then call Voicemail() ?
-Original Message-
From: Phil Finkler [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 12:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 1:04 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG If I
[example]
include = ctx31X
include = ctx3XX
exten = _X.,1,NoOp(this gets executed first for everything)
exten = _X.,2,NoOp(this gets executed second only if ctx31X
or ctx3XX didnt match)
exten = _X.,3,NoOp(this gets executed third for everything)
[ctx31X]
exten = _31X,2,NoOp(this
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 1:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 1:14 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes
-Original Message-
From: Mike [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 1:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG Surely other people have hit the
-Original Message-
From: Mike [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Match a Numer - then continue with
dialplan
Perhaps I can get a clarification
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 2:41 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with,
dialplan
In article
[EMAIL PROTECTED],
Douglas Garstang [EMAIL
-Original Message-
From: Richard Lyman [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Match a Numer - then continue with,
dialplan
Douglas Garstang wrote
I seriously doubt he'd know how to get on the 'Internets'
-Original Message-
From: Doug Crompton [mailto:[EMAIL PROTECTED]
Sent: Wed 12/20/2006 8:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject:RE: [asterisk-users] Re: Match a Numer -
= dynamic
dtmfmode = rfc2833
nat = no
callerid = Douglas Garstang 3254101
secret = password
mohsuggest = moh1
[3254102]
type = friend
context = CallStart
username = 3254102
accountcode = 3254102
qualify = yes
canreinvite = no
host = dynamic
dtmfmode = rfc2833
nat = no
callerid = Douglas Garstang
Anyone know if there's a way to match a dialplan extension, execute some code,
say set a variable, and then continue with the dialplan?
I want to set a variable when the dialplan flows beyond a certain context. This
would be a great feature.
Doug.
]
include = coo1_OnNet
include = coo2_OnNet
[syst_OffNet]
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],180,tr)
~
-Original Message-
From: Douglas Garstang
Sent: Tuesday, December 19, 2006 2:46 PM
To: Asterisk Users Mailing List - Non
-Original Message-
From: David Thomas [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 3:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Match a Numer - then continue with
dialplan
On 12/19/06, Douglas Garstang [EMAIL
-Original Message-
From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 4:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Match a Numer - then continue with
dialplan
Please correct me if I'm
:Re: [asterisk-users] Match a Numer - then continue with dialplan
Douglas Garstang wrote:
I just know someone is going to ask 'why would you ever want to do that?'.
Here's my answer.
We have two companies, each with a dialplan similar to what's below. In the
event that the number being
I'm not sure that any solution with the MySQL dialplan command is going to be
ideal. You also can't nest your queries, ie the connectid/result id seems to
only be good for one resultset at a time... try doing something like
findme/followme with that!
Doug.
-Original Message-
From:
From the Asterisk console:
sip notify polycom-check-cfg ipaddr
Or you might have to switch the polycom-check-cfg and the ip. I forget the
order. You also need to make sure that the phone has alwaysreboot=1 in the
sip.cfg xml file.
Doug.
-Original Message-
From: Klaverstyn, David C
Scenario:
A call is sent from one Asterisk system to another with IAX. The remote
Asterisk system runs the Queue application, which then starts to play a
different music on hold class then the standard 'default'. The console on this
system displays:
-- Executing
Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat'
soft-key to work? When you change the status in this way, the phone does not
send any communication to Asterisk, and it seems to have no effect in incoming
calls. So... what's it for?
Doug
13, 2006 9:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom MyStat
On 12/13/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat'
soft-key to work? When you change
Anyone seen this...? Is it a known issue?
I'd file a bug, but we're on 1.2.9.13, and every time I file a bug and it isn't
against the latest code I get given crap for it. Given that most of the time
you don't know HOW to reproduce a problem on the latest code anyway, not
accepting bugs from
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