days some limit is better
then nothing.
When we found the same problem we had a call that was stuck open for 20
days. The call was stuck in a conference and was sending the on-hold music,
which is what kept it open.
Hope that helps.
Cullin J. Wible
-Original Message-
From: [EMAIL
There is nothing in the GPL that prohibits you from selling the software
(RedHat Software). There is also nothing stops a sales person from denying
it.
They must provide a copy of the GPL and they must give you the source code
and related modifications if you ask (not sure if you have). There are
This is not a SIP issue, but a problem with your configuration.
We have all hard phones register/authenticate with their MAC address as the
user/peer name. Soft phones use user id's that correspond to the person. We
then have our dialplan ring the appropriate devices (soft or hard) depending
on
externnotify=/opt/asterisk-1.2.7.1/sbin/vm-audio
vm-audio uses 'sox -e' to determine how much to scale by without clipping
and then
Then 'sox -v' to scale the sound file.
This happens after the email message is sent, but by changing the order of a
few lines in the app_voicemail.c program you can
such as responding to the report port (instaed of
5060), etc (asterisk standard NAT settigns) will do all that you need.
After spending lots of time with all of this: If you're running STUN you're
trying too hard.
Cullin J. Wible
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
- Non-Commercial Discussion
Subject: Re: [asterisk-users] 1.2.10 - g726 Issues
Cullin J. Wible wrote:
I have hard that 1.2.10 has issues with voice quality through g726.
Can anyone provide any feedback or point me in the right direction
about the current status of this problem?
Been using
.
Cullin J. Wible wrote:
Yeah, that's exactly the problem that I am having here (also switched
to
g729 and gsm).
However, Teliax has told me that the g726 issue is with the * 1.2.10
release and as a result not an issue with their service. I'm not
entirely convinced, but since we also use
I have hard that
1.2.10 has issues with voice quality through g726. Can anyone provide any
feedback or point me in the right direction about the current status of this
problem?
Thanks,
Cullin
___
--Bandwidth and Colocation provided by
as it is probably the preferred solution.
Cullin J. Wible
Co-Founder CTO
Email Data Source, Inc.
212-514-8900 x1006
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins
Sent: Wednesday, June 28, 2006 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial
We run them with 1 call per line, but when we first set them up they would
do 8. The problem was switching between calls on a single line. At that
time, however, the phone did not return busy and allowed the calls to stack
up.
This is set in the XML configuration files.
Cullin
-Original
Polycom phones support STUN - that should solve the issue too.
Cullin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dr. Michael J.
Chudobiak
Sent: Wednesday, June 28, 2006 1:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Because: usg(2)[EMAIL PROTECTED] causes the app to exit with a non-zero status,
Because: [EMAIL PROTECTED]|g(2) causes asterisk to hard crash.
And trying to use g2 in either case doesn't work either.
Cullin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
We did it by comment out a number of lines in the code and then re-compiled
just that module.
We also did the same for the company directory.
Other then that I'm not sure if there's much you can do.
Cullin J. Wible
Co-Founder CTO
Email Data Source, Inc.
212-514-8900 x1006
-Original
In voicemail.conf:
externnotify=/opt/asterisk-1.2.7.1/sbin/vm-audio
The attached script should increase as much as possible without clipping.
Cheers,
Cullin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Technical
Support
Sent: Tuesday, June
/remote users: setup STUN, and use a SPA-1001. For a corporate
setting I highly recommend the Polycom phones.
Cheers,
Cullin J. Wible
Co-Founder CTO
Email Data Source, Inc.
212-514-8900 x1006
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Crompton
have use the local channels to acheive the
functionality that you are looking for as outlined below. It results in a number
of macro's being run in parallel, but as long as you have enough horse power it
shouldn't be a problem.
Hope that helps.
Cullin J.
Wible
Co-Founder
CTO
Email Data
Use Teliax - http://www.teliax.com/
Cullin J. Wible
Co-Founder CTO
Email Data Source, Inc.
212-514-8900 x1006
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arnaud
Sent: Monday, June 19, 2006 7:48 PM
To: Asterisk Users Mailing List - Non
Not that this is an ideal situation, but can you wrap the php code in a
shell script and trap the HUP signal?
Just a thought.
Cullin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Areski K
Sent: Wednesday, November 09, 2005 1:20 PM
To: Asterisk Users
, but the answer after about a week of
research and test accounts was to use Teliax.
Hope that helps.
Cullin J. Wible
President CEO
Algorim Technologies, LLC
212-535-3238 x102
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Piotr A.
Sygula
Sent
We used to run a conference server on a PII 400Mhz with 512MB of RAM. We had
2 separate conference rooms with 15 users each (30 simultaneous) calls with
no problem.
We have since upgraded it to a P4 2.0Ghz with 1GB of RAM (just because it
was getting old) and it still works just fine with even
1) Create the following in your dialplan:
exten = 100,1,VoiceMailMain()
2) Set their password to 1234. They can change it in the voicemail menu.
3) See: Getting MWI on Polycom Phones to work with Asterisk
http://www.voip-info.org/wiki-Getting+MWI+on+Polycom+Phones+to+work+with+Ast
erisk
I
A few comments:
1) We are using quite a few SoundPoint 300 and 501's with no problem.
2) Your intermittent ring issues sounds like what we saw when we tested the
phones through NAT (which doesn't really work at all despite what the
documentation says).
3) We also upgraded to the latest boot
You must use the 't' 'T' options in the Dial() command when placing calls to
and from the device.
We had extensions that were combinations of SIP and IAX devices and didn't
want/need this behavior on all of our devices so we setup our extensions
with something as follows:
Exten =
I had the same problem in 1.0.9. We fixed it by moving the [zonemessages]
section above the [general] section so that it gets processed first.
Cullin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Frank
Tarczynski
Sent: Monday, August 01, 2005 6:53 PM
It is my understanding that the purpose digitmap is to determine when the
phone should transmit the digits entered to the server. I do not believe
that it has any method for changing the dialstring.
However, you could place the Polycom phones in their own context which would
perform this mangling
Use the r option in the Dial() command.
Cullin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nick
Kartsioukas
Sent: Tuesday, July 26, 2005 4:57 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Generate ring while waiting for SIP
I think you could accomplish this with EAGI or the manager interface. You
should also read:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
On a side note, we spent lots of time when we setup our Asterisk system
dealing with the answer detection for PSTN calls. We use Teliax
I always have two as well - not sure why though.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Billy Dunn
Sent: Tuesday, July 26, 2005 6:25 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] mpg123 - two processes
Does everyone have two
From: Cullin J. Wible
[mailto:[EMAIL PROTECTED] Sent: Monday, July 25, 2005 3:36
PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion';
'asterisk-dev@lists.digium.com'Cc: 'David Rule'Subject:
Voicemail Send Message (Options 3, 5) Patch
We run Asterisk
1.0.9 with multiple
1) You could use asterisk realtime and a mysql database.
2) You could use an asterisk database and allow users to set call forwarding
by calling an extension.
3) You could write some scripts to use an external database (what we did)
and either allow users to update their forwarding options via a
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cullin J.
Wible
Sent: Thursday, July 14, 2005 11:58 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: 'David Rule'
Subject: RE: [Asterisk-Users] Unable to call certain 800 numbers through
Teliax
After all
After all of your feedback and a discussion at Teliax we have fixed this
issues.
It appears that when dialing a PSTN number, using the 'r' option is really
unnecessary.
Furthermore, some IAX clients and older phones (e.g. Cisco 20 VIP) require
us to Answer() the call before dialing the PSTN
While I have never done this, it appears that you could use the agents.conf
to allow people to login to an extension and have calls forwarded to their
current phone. The dialplan would then reference the Agent/XXX rather then
the device they are working at.
Hope that points you in the right
, Vertical Networks,
VocalData,
Alcatel and 3COM. For more information on Polycom supported IP
Communications platforms--
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cullin J.
Wible
Sent: Tuesday, July 12, 2005 7:55 PM
To: 'Asterisk Users
We are unable to call certain 800 numbers through Teliax but I thought I
would post this here and see if anyone else had the same problem with either
Teliax or other carriers.
The 800 numbers causing problems pick-up the call right away and are in the
US - American Airlines (8004337300) and
We just purchased 4 of the Polycom SoundPoint 301's.
We are very happy with them so far.
Cullin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, July 12, 2005 8:22 PM
To: Asterisk Users Mailing List - Non-Commercial
Add the r parameter to the end of the Dial() statement.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Wong
Sent: Tuesday, July 12, 2005 10:38 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] NO calling tone
Hi,
When I make a call
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