Hi Bruce,
We have build an Invoicing module (ICTInovice) for Elastix. It is Free, Open
Source, Generate PDF Invoices, and can mail invoices to clients!
You can download it from http://sourceforge.net/projects/ictinvoice/
http://sourceforge.net/projects/ictinvoice/Note: Currently ICTInvoice only
Hi,
I am trying to dial a registered user via his IP:Port mechanism, but problem
is that the audio data is not reaching to dialed user. here is the scenario.
caller and callee both are registered at asterisk server. asterisk server is
on public ip so no port forwarding and natting necessary
Hi,
On 08-02-2010 20:55, Gordon Henderson wrote:
I generated invoices with PHP code - it uses a LaTeX template which it
fills in the gaps, then feeds it through LaTeX and dvi2pdf to generate
PDFs.
Bit of a geek solution though.
well, then i must be a geek too, because i also decided to
Hi all,
I have
exten = _X.,1,Macro(dundi-priv,${EXTEN})
exten = _X.,2,DIAL(CAPI/contr1/${EXTEN})
Now my problem is, that after hanging up a call, the call is instantly
re-established using the h-extension which is almost a loop.
I am sure this is a stupid question, but what am I
Hello list,
scrambling different pieces of info together I've come with the following :
I want to have my register = statements in a MySQL-database, so I've
made the following table.
table ast_config :
id 1
cat_metric 0
var_metric 0
commented 0
filename sip.conf
category general
Hi!
Is there a way to change the mappings of disconnect reasons to certain
SIP messages? E.G. I need to change the mapping for SIP 402 Payment
Required from 16 (normal termination) like it is in 1.4.24 to 21
(call rejected) as defined in RFC 3398.
* if you think the mapping is wrong, then
Is asterisk and the SIP device behind the same router?
Most routers will not redirect internal NAT requests. So that if you
are trying to have port forwarding done but the request and the
forwarding destination are on the same interface it won't work.
On 8/3/10, Nasir Javaid
Hi!
Question 1 :
[Aug 2 13:56:57] Capabilities: us - 0x90a (gsm|alaw|g726|g729), peer -
audio=0x808 (alaw|g726)/video=0x0 (nothing), combined - 0x808 (alaw|g726)
why is combined alaw|g726 and not g726|alaw (reverse) ??
Guess: Here the order presented has no meaning for the order of codec
In most cases wait(.5) will do. I would not recommend using
answer(2000) as that answers the channel, which means you start
getting billed.
On 8/2/10, Peder pe...@networkoblivion.com wrote:
I am using T1's and didn't think the spill would take that long.
PRI no, EM yes.
Some PRI take that
They seem to have taken over manufacture of cards Digium has discontinued.
I have used several of the TE110 card with success and they are identical
John Novack
asteriskguru asteriskguru wrote:
hi,
I am using this card and IP phone about 6 months. There is no issues
at all.
Installation
Hello Philipp,
thank you for your answer.
On 08/03/2010 01:21 PM, Philipp von Klitzing wrote:
Question 3 :
How can I get g726 as first preferred codec ??
Which Asterisk version are you using?
Using Asterisk 1.4.30
* check if you have disallow/allow settings in the [general]
We all know what are you trying to do, and it is not possible to do, but it
is very impolite and annoying to repost it every few days as a new post with
a slightly different subject. Nobody else does it, and you too please avoid
doing it.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-08-03
I know someone who uses a billing solution called 'freeside', and is happy
with it. Personally I developed my own solution because none could satisfy
my needs.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-08-03 2:34 AM, ad...@3a.hu wrote:
Hi,
On 08-02-2010 20:55, Gordon Henderson wrote:
I agree but the mentioned software is not opensource.
My conditions clearly included opensource.
On Tue, Aug 3, 2010 at 12:35 AM, Nick Brown n...@ipera.com.au wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
Oh, you seem to be right on. It's actually an install of Elastix. I will be
testing this for sure. Hope it doesn't do any damages though.
I guess the installation material is inside the tar ball?
Thanks
On Tue, Aug 3, 2010 at 2:01 AM, Nasir Iqbal na...@ictinnovations.comwrote:
Hi Bruce,
We
Yep, I seen that. That is probably the closet thing but looking at he
interface it makes me not try to install it. Maybe too complicated. I
wouldn't want to send customer the whole CDRs but rather a nice Bill like
the telco sends out.
I am currently toying with NCH Invoicing. Those guys make a
http://gigaom.com/2010/08/03/2600hz-project/
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Most customers I have dealt with, including myself, want CDRs with the bill
via online access, or emailed as an attachment (if not too large).
How can they reconcile their bill without call details?
One vendor I used would send a DVD each month with CDRs because we were
averaging ~15,000 -
I wanted the same, and so wrote my own. There is none free for this purpose.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-08-03 9:32 AM, bruce bruce bruceb...@gmail.com wrote:
Yep, I seen that. That is probably the closet thing but looking at he
interface it makes me not try to install it.
I'm still working on it, but I am using a2billing and making
modifications to some of the PHP code. I modified the layouts of their
default invoices and and added PDF creation using dompdf
(http://code.google.com/p/dompdf/downloads/list).
-b
On 08/03/2010 09:41 AM, Zeeshan Zakaria wrote:
Also:
There are at least two implementations of the g726 codec, i.e. g726 and
g726aal2. For this also look at the g726nonstandard setting in sip.conf.
It is quite possible that your problem is here.
For quick testing to see if the codec works at all: Configure your phones
to do g726 only (so
On Tue, Aug 3, 2010 at 6:24 PM, bruce bruce bruceb...@gmail.com wrote:
Oh, you seem to be right on. It's actually an install of Elastix. I will be
testing this for sure. Hope it doesn't do any damages though.
I guess the installation material is inside the tar ball?
It is very easy to
Hi!
In the [general] section of sip.conf I have :
disallow=all
allow=g726
allow=alaw
allow=g729
allow=gsm
So change the order there and see what happens.
* look at the variable SIP_CODEC for the inbound (first) call leg, and
in Asterisk 1.8 (or 1.6.2?) also at SIP_CODEC_OUTBOUND
On Tuesday 03 August 2010 06:21:23 Philipp von Klitzing wrote:
Is there a way to change the mappings of disconnect reasons to certain
SIP messages? E.G. I need to change the mapping for SIP 402 Payment
Required from 16 (normal termination) like it is in 1.4.24 to 21
(call rejected) as
I've been replacing an old Toshiba DK switch with an Asterisk solution. I'm
needing a solution for fax machines that works as well as a POTS line from
my carrier. If the POTS line is the solution, I'll keep it, but I'd rather
move away from that.
Here's what I'm thinking...will it work?
I
On Tue, Aug 03, 2010 at 02:28:15PM +0100, Alan Lord (News) wrote:
http://gigaom.com/2010/08/03/2600hz-project/
So practically FreePBX V3 was renmed 2600Hz / BlueBox ?
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
On 08/03/2010 10:48 AM, Joel Maslak wrote:
I've been replacing an old Toshiba DK switch with an Asterisk solution. I'm
needing a solution for fax machines that works as well as a POTS line from
my carrier. If the POTS line is the solution, I'll keep it, but I'd rather
move away from that.
Kyle Kienapfel doctor.whom at gmail.com writes:
NOTICE.* .*: Registration from '.*' failed for 'HOST' -
ACL error \(permit/deny\)
I don't see slashes in front of the brackets on what you posted to the
mailing list. I'm posting my config to see if the mailing list mangles
it
Isn't that trademarked? :P
On Tue, Aug 3, 2010 at 9:28 AM, Alan Lord (News) alansli...@gmail.com wrote:
http://gigaom.com/2010/08/03/2600hz-project/
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http://www.theopenlearningcentre.com
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- Joel Maslak jmas...@antelope.net wrote:
I've been replacing an old Toshiba DK switch with an Asterisk solution. I'm
needing a solution for fax machines that works as well as a POTS line from my
carrier. If the POTS line is the solution, I'll keep it, but I'd rather move
away from
Joel Maslak wrote:
So...will this work?
It will work very well, I have two installations with ADIT600s
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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/attachments/20100803/08ce42aa/attachment-0001.htm
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Message: 17
Date: Tue, 3 Aug 2010 09:11:07 -0400
From: Zeeshan Zakaria zisha...@gmail.com
Subject: Re: [asterisk-users] RTP stream not passing through router
withport forwarding
To: Asterisk Users Mailing
wow which idiot decided to associate this project with call phreaking.
really dumb move.
Shame this is where AMP has ended up based on where it started from.
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
+1-212-203-4357 New York
+61-2-9016-5642 (Sydney in-dial).
On 10-08-02 02:26 PM, bruce bruce wrote:
Hi Everyone,
Sorry, if it's not directly related to Asterisk. Some of people on this
list might have PBX deployed for their clients. What software do you use
to invoice them so the invoice looks like a proper telecom invoice maybe?
Prefer:
On Tue, 2010-08-03 at 09:17 -0400, bruce bruce wrote:
I agree but the mentioned software is not opensource.
My conditions clearly included opensource.
No, your prefer listed opensource. If you had said requirement I
wouldn't have suggested it.
j
On Tue, Aug 3, 2010 at 12:35 AM, Nick
I am having a problem with asterisk voice mail messages that seems to be
intermittent. Though the problem occurs most of the time, on rare occasions
it will work fine - rare enough that I can't pin down what it is that works.
The problem is that voice mail message get played back garbled.
Any body experiencing incoming calls into a sip-agent which is
already registered into a real-time queue with Asterisk, having an active
call?
We are using Asterisk Version 1.4.21.2
Al
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bobby Larson
Subject: [asterisk-users] Garbled messages - format_wav_gsm.c:414
wav_read:Short read (60) (No such file or directory)!
The problem is that voice mail message get played
Hi Bruce,
Did you ever get a working solution and confirm the underlying issue ? I am
having the same issue on a set of phones, my next step is replacing the
router, but I was wondering if you found something else.
Regards,
Mike
From: asterisk-users-boun...@lists.digium.com
Hi,
I have a question about the use of Confbridge.
In my dialplan, I allow a caller to reserve a conference room by choosing a room
number and a password. Thus, any other caller who would like to join the same
room can do it only if he knows the password. It works fine, but I need to free
the
Hello Al,
On 03.08.2010 20:06, Alvaro Ramirez wrote:
Any body experiencing incoming calls into a sip-agent which is
already registered into a real-time queue with Asterisk, having an
active call?
We are using Asterisk Version 1.4.21.2
Yes - see
Hi All,
I'm connecting to my carrier which requires setting of outboundproxy. There
has been few cases where the proxy server failed due to network issues and
required us to use a secondary one. Is there a timeout or qualify setting
for outboundproxy setting in sip.conf?
I do appreciate if
Thank you doctor whom,
It is working for me now.
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New to Asterisk? Join us for a live introductory webinar every Thurs:
Hi,
I can't figure out what syntax to use with the Dial() M parameter
for the AEL parser to interpret properly. Creating an AEL
macro named macro-screen() partly works as a hack, but it must
not turn into a gosub properly, so I get warnings about the return;.
Dial(...,tgM(screen)) with the
No its a split
FreePBX is still the same, V3 is still the same, this is a fork from some guys
who had got involved (or maybe paid some money)
Cheers Duncan
On 4/08/2010, at 2:56 AM, Tzafrir Cohen wrote:
On Tue, Aug 03, 2010 at 02:28:15PM +0100, Alan Lord (News) wrote:
On Tue, Aug 3, 2010 at 2:26 PM, Duncan Turnbull dun...@e-simple.co.nz wrote:
FreePBX is still the same, V3 is still the same, this is a fork from some
guys who had got involved (or maybe paid some money)
That is how I read the announcement.
--
Paul Belanger | dCAP
Polybeacon | Consultant
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark G. Thomas
Subject: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2
Hi,
I can't figure out what syntax to use with the Dial() M parameter
for the AEL parser to interpret properly.
On Tuesday 03 August 2010 13:19:11 Mark G. Thomas wrote:
I can't figure out what syntax to use with the Dial() M parameter
for the AEL parser to interpret properly. Creating an AEL
macro named macro-screen() partly works as a hack, but it must
not turn into a gosub properly, so I get warnings
On 3 August 2010 19:54, Paul Belanger paul.belan...@polybeacon.com wrote:
On Tue, Aug 3, 2010 at 2:26 PM, Duncan Turnbull dun...@e-simple.co.nz
wrote:
FreePBX is still the same, V3 is still the same, this is a fork from some
guys who had got involved (or maybe paid some money)
That is how
Hi,
On Tue, Aug 03, 2010 at 01:49:11PM -0500, Tilghman Lesher wrote:
On Tuesday 03 August 2010 13:19:11 Mark G. Thomas wrote:
I can't figure out what syntax to use with the Dial() M parameter
for the AEL parser to interpret properly. Creating an AEL
macro named macro-screen() partly works
AEL is very simple and the instructions on voip-info.org are enough to learn
it. In fact I can't understand how can one write complex dial plans not
using AEL, you simply can't do it using standard format used in
extensions.conf.
As for the tutorials, there is no specific website for them as per
AEL is very simple and the instructions on voip-info.org are enough to learn
it. In fact I can't understand how can one write complex dial plans not using
AEL, you simply can't do it using standard format used in extensions.conf.
As for the tutorials, there is no specific website for them as
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
unsero...@aol.com
Subject: Re: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2
AEL is very simple and the instructions on voip-info.org are enough to
learn it. In fact I can't
I am trying to add an Asterisk box to an Iwatsu ECS (Software Version
7.0 R.01) hopefully without using a physical T1/E1 card. Internally the
SIP works fine, it is dialling an outside line that is giving me
difficulties. One way that I think it might be possible is for an
outbound call to
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
jeremy.hellst...@synovate.com
Subject: [asterisk-users] Using SIP to dial extension that will give
anoutside line
You could try this:
; use lwatsu line
Exten =
On Tue, 2010-08-03 at 16:04 -0500, Danny Nicholas wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
jeremy.hellst...@synovate.com
Subject: [asterisk-users] Using SIP to dial extension that will give
anoutside line
You
We should be hearing more on this from Darren either this Friday or next on VUC.
http://vuc.me
/r
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Please note that I don't claim myself a guru, just happened to be working
with Asterisk for some good number of years, so probably know some stuff
better than others.
As for the number of lines, 1800 lines will come down to 1000 lines using
AEL but not the opposite.
When I'll be back home,
Hi Mike,
I am putting the phones on AC Adapter now as I am suspecting the Linksys POE
switch. Once that test is done and if problem still presists, I will be
enabling DHCPMasq and also set the SIP registration time to 1 second on the
phone UI.
-Bruce
On Tue, Aug 3, 2010 at 1:13 PM, Mike
On Tue, Aug 3, 2010 at 1:30 PM, Mark G. Thomas m...@misty.com wrote:
I didn't know there was a U option. I don't see any mention of it
on the voip-info.org wiki or other Dial() documentation, but didn't
check for new options in the built in documentation until just now.
I updated the dial
-Original Message-
From: asterisk-users-boun...@lists.digium.com on behalf of Carlos Chavez
Sent: Tue 8/3/2010 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Using SIP to dial extension that will give
anoutside line
On Tue,
On Tue, Aug 3, 2010 at 4:49 PM, jeremy.hellst...@synovate.com wrote:
Unfortunately it is only the Iwatsu IP phones that grab the open line @
3001 currently, the softphones do not. I might try programming the
extension and see if I can get a response that way.
Mostly what I am seeing is
Any one can help please?
I go over the problem again.
Let's say I have the following peers
[trunk1]
host=sip.provider.com
outboundproxy=sbc1.probider.com
type=peer
[trunk2]
host=sip.provider.com
outboundproxy=sbc2.provider.com
type=peer
Let's say I call by SIP/trunk1/number and the proxy
On Tue, Aug 3, 2010 at 9:58 PM, Abeed Saleh abeedsa...@gmail.com wrote:
Let's say I call by SIP/trunk1/number and the proxy server is down, is there
a way to get CHANUNAVAIL?
*CLI core show application Dial
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Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com |
Hi Paul,
Thank you for your reply. Unfortunately the timeout parameter will not do
the job for me. I need something equivalent to qualify to monitor the
outboundproxy.
Best
On Tue, Aug 3, 2010 at 7:25 PM, Paul Belanger
paul.belan...@polybeacon.comwrote:
On Tue, Aug 3, 2010 at 9:58 PM, Abeed
Hi,
Can someone help me formulate MySQL Query(s) which will help me extract the
following details for a given DID (date range can be excluded for
simplicity).
Date-Time
DNID (I am recording this is `userfield`)
CLID
time-1 (when call was received)
time-2 (when call was answered by agent)
time-3
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