If you set invite=insecure,port in the general section of sip.conf and do
not mention invite settings in the user/peer section i think it will work
like you want. you have to test it first coz i havent.
On 2/23/07, dima [EMAIL PROTECTED] wrote:
Hello, everyone.
I'm having a small problem when
hi guy, i have a problem, i have an sellvoip account and i want
configure asterisk for outbound calls.
this is my sip.conf
register = X00:[EMAIL PROTECTED] ; this is one of the
sellvoip server
[sellvoip_out]
type=friend
secret=PassWord
username=XX00
host=70.42.34.200
22 feb 2007 kl. 23.40 skrev Philipp Kempgen:
Olle E Johansson wrote:
22 feb 2007 kl. 19.34 skrev Philipp Kempgen:
I thought it might be useful to be able to ask Asterisk for the
current SIP CSeq through the Manager API in order to send your
own SIP messages during a call outside of
23 feb 2007 kl. 06.52 skrev Yuan LIU:
Quite a few documents, including voip-info, make reference to this
term. (e.g., First, You need trunk version of Asterisk.) But I
can't seem to find anything that defines this. In SVN, trunk
simply refers to the main body of code. Can someone
Hey,
We have asterisk 1.2.4 (old I know) with a couple of snom
phones, a couple of grandstream phones and around 65 philips
dect stations.
Now the problem:
All calls do peer to peer RTP except the calls from dect
station to dect station.
snom to dect or dect to snom do peer to peer.
So the sip
Dear Facundo,
http://www.asterisksupport.org/tiki-index.php
You can create spanish pages on this tiki.
Rehan
Date sent: Thu, 22 Feb 2007 19:13:55 -0300
From: Facundo Ameal [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
This is my solution to manage an Agi script as a queue member.
http://www.chiese.tn.it/index.php?sezione=softwareoperazione=dettaglioid=14
The script can be simply adapted to manage may queues, many agi script.
Bye nik
___
--Bandwidth and Colocation
Quoting Charles Wang [EMAIL PROTECTED]:
Dear Phil,
The extension 'h' was a great idea although I still got the error
exited non-zero.
You will. Dial() always exits non-zero on hangup.
--
Phil Reynolds
o mail: [EMAIL PROTECTED]
|L_ \ / Web: http://www.tinsleyviaduct.com/phil/
(_)-
In article [EMAIL PROTECTED], lenz [EMAIL PROTECTED] wrote:
Hello list,
I have prepared a small tutorial today that deals with how to avoid
Asterisk rebuilding DTMF tones when using it to connect industial
appliances that use DTMF. You can find it at:
I heard from a Voxbone executive that they are opening a new NOC near Los
Angeles and probably hiring a few voip support engineers.
If anyone is interested feel free to contact them through their website
Cheers,
Robert
Hi list!
I have an Asterisk server (1.2.14) connected to a E1 line via a TE410P, and
some
PAP2NA connected to it. The PAP2 DTMF configurations is set to INFO and
Asterisk
to INFO too. At first, is INFO method different from RFC2833??
Well, I have two problems. The first is that when I place a
Hi,
In older versions of asterisk I used to be able to use
incominglimit=1 to effectively disable call waiting on a specific
SIP channel (Where broken phones do not allow this on the handset
itself)
In 1.2.x this became call-limit=1, but this prevents the phone from
opening a 2nd line in order
Hi,
Just need to confirm whether dial() command provided options
h: Allow the callee to hang up by dialing *
H: Allow the caller to hang up by dialing *
work for SIP channels as well ?
-ag
___
--Bandwidth and Colocation provided by Easynews.com --
I use Zaptel with PRI. Can I safely install BRIStuff to get this ability
and still not break anything?
On 2/22/07, Sune Kloppenborg Jeppesen [EMAIL PROTECTED] wrote:
On Thursday 22 February 2007 23:01, Norbert Zawodsky wrote:
This sounds interesting. If it's not too complicated for you
On 08:13, Fri 23 Feb 07, Matt wrote:
I use Zaptel with PRI. Can I safely install BRIStuff to get this ability
and still not break anything?
Sure.
bristuff will install a patched zaptel, but I never noticed
it broke zaptel stuff that was working with stock zaptel.
--
Michiel van Baak
[EMAIL
On 18:06, Fri 23 Feb 07, ast guy wrote:
Hi,
Just need to confirm whether dial() command provided options
h: Allow the callee to hang up by dialing *
H: Allow the caller to hang up by dialing *
work for SIP channels as well ?
yes. It's a function in asterisk call thingie, not in the
sip
On Fri, 2007-02-23 at 00:25 +0100, Sune Kloppenborg Jeppesen wrote:
On Thursday 22 February 2007 23:01, Norbert Zawodsky wrote:
This sounds interesting. If it's not too complicated for you
This should get you going:
in extensions.conf:
[macro-F_Toggle_status] ; $ARG1 db family $ARG2
And presumably you can roll back to the regular zaptel if need be? Does
bristuff install it's OWN patched zaptel? Or do you supply the zaptel
source code for it to patch?
On 2/23/07, Michiel van Baak [EMAIL PROTECTED] wrote:
On 08:13, Fri 23 Feb 07, Matt wrote:
I use Zaptel with PRI. Can I
What you have in your sip.conf only handles inbound calls. You need to
add something like the following to your extensions.conf to enable
outbound calls:
Exten = _1NXXNXX, 1,
Dial(IAX2/XX00:[EMAIL PROTECTED]/${EXTEN})
Exten = _1NXXNXX, 2,
Hello,
I'm interested too in analyzer/statistics/billing system. Can we develop
together something simple? What scripts do you recomand me?
Thank you,
Jonson.
On 2/22/07, nik600 [EMAIL PROTECTED] wrote:
I am planning to develop an open source (GPL) queue statistic/analyzer.
Can i use that
In the last months i've developed a web application for the use of an
asterisk call center.
Yuo can
- make calls from a web interface
- login/logout in queue
- view members logged in a queue
- view callers queued in a queue
- pickup a callers from a queue
I am planning to add new features
-
I do believe it is that chipset.
The person placing the call from the AG-188 does not hear a ring.
--Mike
Message: 8
Date: Fri, 23 Feb 2007 01:21:52 +
From: Thomas Kenyon [EMAIL PROTECTED]
Subject: Re: [asterisk-users] AG-188
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello Angel.
Did you solve this issue?
I have the same problem.
Thanks,
José
El jue, 01-02-2007 a las 01:15 -0800, Angel Heart escribió:
Hi,
I am using FXO 8 Ports AudioCodes. Asterisk is able to accept incoming
outgoing calls. However, I noticed that the caller ID of the caller
Lacy Moore - Aspendora wrote:
On 2/21/07, Stephen Bosch [EMAIL PROTECTED] wrote:
My point is that if it's going to involve rebuilding a kernel to support
IO-APIC, then I'd just as soon build from the ground up.
And my point is that this is the Asterisk Users mail list, not the
Trixbox list.
Hi,
I'm trying to make ooh323 works with one asterisk box running 1.2.15
version.
I can ring from a h.323 to SIP and SIP to H.323, but when the call is
finished when the phone is answered.
This is the log when I call from the H.323 device to a SIP device:
Feb 23 10:57:32 VERBOSE[6096]
We're having a lot of D channel problems with the pci-e on HP servers.
Going to PCI fixed the problem. Sangoma is aware of the problem and is
using one of our servers to work toward a solution.
-Jeremy
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I fogot, the H.323 device is one Antek networks INC with two fxo ports.
Regards,
On Fri, 2007-02-23 at 11:07 -0500, Guillermo Salas M. wrote:
Hi,
I'm trying to make ooh323 works with one asterisk box running 1.2.15
version.
I can ring from a h.323 to SIP and SIP to H.323, but when the
How does one answer more than 100% of the calls in less than 60 seconds?
techsupp has 0 calls (max 20) in 'leastrecent' strategy (104s
holdtime),
W:0, C:3, A:2, SL:166.7% within 60s
___
--Bandwidth and Colocation provided by Easynews.com --
On 2/23/07, Stephen Bosch [EMAIL PROTECTED] wrote:
I saw your point, and I disagree.
Trixbox is what it is, and it is built on Asterisk. Without Asterisk,
there is no Trixbox. Moreover, as long as the Trixbox forums and
documentation are as weak as they are, you can expect to see Trixbox
I know this is not a Polycom support forum, but I also know there are a lot
of you with a great deal of Polycom experience.
Is there anyway to remove the Attended Transfer but keep the Blind
transfer? Or better yet, just swap the two soft buttons locations?
I know you can remap the Hard buttons,
In later 1.6.x firmwares there is a config option for allow transfer on
proceeding that basically allows you to do a blind transfer by just
hitting the transfer key again rather than having to select Blind.
Shawn Kelley wrote:
I know this is not a Polycom support forum, but I also know there
Olle E Johansson wrote:
22 feb 2007 kl. 23.40 skrev Philipp Kempgen:
Olle E Johansson wrote:
22 feb 2007 kl. 19.34 skrev Philipp Kempgen:
I thought it might be useful to be able to ask Asterisk for the
current SIP CSeq through the Manager API in order to send your
own SIP messages during a
Hi,
Do you think it could have been done with another T1/E1Asterisk box between
the Nortel PBX and the other Asterisk server ?
Which features would you then loose or gain, given current status of QSIG
support in Asterisk ?
Regards
___
--Bandwidth and
Hi,
I have just setup inbound SIP and wonder if somebody would be so kind as to
test that it works for me, and that my
firewall is setup okay.
sip:[EMAIL PROTECTED]
Thank you
--
--[ UxBoD ]--
// PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import
// Fingerprint: 543A E778 7F2D
Yeah but I think the caller ID issue still remains.
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Friday, February 23, 2007 11:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Solved... installed chan_oh323 :)
http://www.ecualug.org/?q=2006/02/28/comos/asterisk_1_2_4_agregando_soporte_para_el_protocolo_h_323
I don't know why ooh323 does not work.
Regards,
On Fri, 2007-02-23 at 11:21 -0500, Guillermo Salas M. wrote:
I fogot, the H.323 device is one Antek networks
Problem solved and posted below.
I may have found out at lease two things that may help others.
Especial thanks to Eric ManPower, Benny Amorsen, Luan LIU, Paul Hales,
Pavel Jezek and a _lot_ of research on the web.
The problem was separating the contexts for incoming VOIP calls
from the
I'm trying to buy the cisco firmware update but it seems that i cannot
order online because I bought my 7970 on ebay. Is there any other
chance to get this update? ... anyone can make me a favour and send it
to me by email?
thank you
___
--Bandwidth
We had an issue, and I know others had posted the same on the list.
Scenario:
Polycom phone user sets call forward to a toll free number(in our case)
Call arrives for the phone, the phone notifys asterisk, asterisk
dials new number.
Telco drops call. But if you dial direct to the number it
Hey all,
This should be an easy one. I have a few different queues and wanted to
set up a standard macro to handle them, so I can shrink the dial plan
down and stop having so much redundancy. But when I try to use it, i get
a no answer.
Here's what does work (non macro):
exten = 5054,1,Answer()
On Fri, 23 Feb 2007 19:09:40 +
--[ UxBoD ]-- [EMAIL PROTECTED] wrote:
Hi,
I have just setup inbound SIP and wonder if somebody would be so kind as to
test that it works for me, and that my
firewall is setup okay.
sip:[EMAIL PROTECTED]
Thank you
Thank you very much to westcomuk
Hi
i install Asterisk can register softphones on clients computers but when i
make a call to a extencion this error apear
Call Failed: not found
in the asterisk machine i do commannd sip show peers and i can see the
clients there
can you help me
thanks
Rob Schall wrote:
Here's the macro I tried to make and use:
[coqueuevm]
The name of the context should be macro-coqueuevm.
The macro- part will automatically be cut by the
Macro() application.
exten = 5054,1,Macro(coqueuevm,itdept,5054)
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 -
That was it. :) Thanks much!
A followup... well, kinda related... And not really a asterisk q.
On the polycom 501 phones... There's those 3 lines that you can setup.
Is it possible to make one of them a shortcut to the queue login/logout
extension?
Rob
Philipp Kempgen wrote:
Rob Schall
Rob Schall wrote:
On the polycom 501 phones... There's those 3 lines that you can setup.
Is it possible to make one of them a shortcut to the queue login/logout
extension?
Haven't used that phone myself but it seems like you need
to add your queue extension (5054) to the phone's directory
and
Philipp Kempgen wrote:
Rob Schall wrote:
On the polycom 501 phones... There's those 3 lines that you can setup.
Is it possible to make one of them a shortcut to the queue login/logout
extension?
Haven't used that phone myself but it seems like you need
to add your queue extension (5054)
I have a bunch of sounds that I've converted into gsm from (Indexed NMS) vox
files. There's only a single utility that I've found that can read and
convert vox files. My conversion process is to use this utility to convert
the index vox file in to a series of wave files and then use sox to convert
Audacity
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
David Ruggles
Sent: Friday, February 23, 2007 4:48 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] GSM cleanup (pops, clicks and static)
I
On Fri, 2007-02-23 at 16:48 -0500, David Ruggles wrote:
I have a bunch of sounds that I've converted into gsm from (Indexed NMS) vox
files. There's only a single utility that I've found that can read and
convert vox files. My conversion process is to use this utility to convert
the index vox
Wavepad ( a windows program ) is MUCH easier to use
John Novack
Robert Augustyn wrote:
Audacity
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
David Ruggles
Sent: Friday, February 23, 2007 4:48 PM
To: 'Asterisk Users Mailing List -
Hi, how i have to do for receive a email with a alert from my voice mail?
My doubt is what I put in serveremail in file voicemail.conf. I think is a
email server, but can be see anyone? I searching one in the internet?
Thanks and sory my english
Hi, i configured Musiconhold and Works, but the sound is very low. I haved
put the volume in the max, but is equal.
I tested to my voice, and the sound is also low.
exten=8000,1,Wait(2)
exten=8000,2,Record(menu:gsm)
exten=8000,3,Wait(2)
exten=8000,4,Playback(menu)
exten=8000,5,Hangup()
Carlos Jerónimo wrote:
Hi, how i have to do for receive a email with a alert from my voice mail?
You need a working installation of sendmail on your server.
Then you append the email address of the users to the mailbox
definitions in voicemail.conf like this:
1234 = 1234,Some User,[EMAIL
Hello all,
I'd like to introduce you to a new feature that we're opening up for all users
on the AsteriskNOW.org website. Anyone with an account can now post a blog on
the front page. This feature will give you the opportunity to post stories
about what you've done with Asterisk and
Hi all, I'm having a problem, with the h extension.
I have an application, when I call it check for the line requested and then
direct the call to a predefined context.
In this context I play a message (the message according to the line called)
and then park the call.
The dialplan does some
On 24/02/07, Tim Connolly [EMAIL PROTECTED] wrote:
How does one answer more than 100% of the calls in less than 60 seconds?
techsupp has 0 calls (max 20) in 'leastrecent' strategy (104s
holdtime),
W:0, C:3, A:2, SL:166.7% within 60s
Probably talking out of my hat (I've never particularly
On Fri, Feb 23, 2007 at 11:33:30PM +0100, Philipp Kempgen wrote:
Carlos Jerónimo wrote:
Hi, how i have to do for receive a email with a alert from my voice mail?
You need a working installation of sendmail
A sendmail, actually. postfix, exim or whatever will also do.
Or even nullmailer or
Hello:
I have some questions regarding using a striped version of asterisk compiled
in a mips32 adsl router (probabilly broadcom 96348R Linux version 2.6.8.1
([EMAIL PROTECTED]) (gcc version 3.4.2)
I would like your comments on this dialplan
(do you think it will work)
[frombooths]
exten =
Thanks Philipp.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
Kempgen
Sent: sexta-feira, 23 de Fevereiro de 2007 22:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voice mail server
Carlos Jerónimo
Thanks Philipp.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
Kempgen
Sent: sexta-feira, 23 de Fevereiro de 2007 22:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voice mail server
Carlos Jerónimo
I really don't get it
From several emails in this list archive, I had clearly understood that
it is important to switch Echo Cancellation off for fax-channels, or
faxing would not work properly.
However, faxing (B410P ISDN bridged to TE410P PSTN) seems to work fine
with EC at 256 taps
Zoilo Gomez wrote:
I really don't get it
From several emails in this list archive, I had clearly understood
that it is important to switch Echo Cancellation off for fax-channels,
or faxing would not work properly.
However, faxing (B410P ISDN bridged to TE410P PSTN) seems to work fine
How do I receive text sent from SendText() application? Asterisk lists text
capability, so SendText() is successful. But I don't see an application to
actually use it.
Yuan Liu
___
--Bandwidth and Colocation provided by Easynews.com --
On Friday 23 February 2007 8:35 pm, Zoilo Gomez wrote:
However, faxing (B410P ISDN bridged to TE410P PSTN) seems to work fine
with EC at 256 taps on the B410P.
Generally speaking all modems (this includes POS machines and faxes) emit a
tone which echo cancellers recognize and disable
Zoilo Gomez wrote:
I really don't get it
From several emails in this list archive, I had clearly understood
that it is important to switch Echo Cancellation off for fax-channels,
or faxing would not work properly.
However, faxing (B410P ISDN bridged to TE410P PSTN) seems to work fine
Do you think it could have been done with another T1/E1Asterisk box between
the Nortel PBX and the other Asterisk server ?
Sorry, I do not understand exactly what you are asking. Do you mean using an
Asterisk with PRI card instead of Cisco? If so, I have no experience with this.
Which
show dialplan keeps showing contexts created by AEL. I tried deleting
/etc/asterisk/extensions.ael but kept getting these messages in the Asterisk
log:
Feb 14 21:39:53 WARNING[6074] pbx_ael.c: Unable to open
'/etc/asterisk/extensions.ael': No such file or directory
Feb 14 21:39:53 WARNING[6074]
It would be something like
noload = pbx_ael.so
in /etc/asterisk/modules.conf
later,
PaulH
On Sat, 2007-02-24 at 16:16 +1100, Eric Bishop wrote:
show dialplan keeps showing contexts created by AEL. I tried
deleting /etc/asterisk/extensions.ael but kept getting these messages
in the
Eric Bishop wrote:
show dialplan keeps showing contexts created by AEL. I tried
deleting /etc/asterisk/extensions.ael but kept getting these messages
in the Asterisk log:
Feb 14 21:39:53 WARNING[6074] pbx_ael.c: Unable to open
'/etc/asterisk/extensions.ael': No such file or directory
Feb 14
Hi
Hi
Is there any accessible ocumentation, ie plain text or html, how to
configure Asterisk. The book
'Asterisk: The Future of Telephony'' is availablly only as and pdf
document and is thus unreadable for a blind user.
Any pointers welcome.
You can still escape from the
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