Probably worth noting that sipgate will close (at least in the U.S.) on Oct.
31:
http://www.besttechie.com/2013/09/13/voip-provider-sipgate-will-close-oct-31/
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
What is your provider seeing? Many providers send re-INVITEs at 15 minutes.
Many firewalls have closed their port before this due to UDP timeouts. I have
a whitepaper that I wrote on this subject; I will see if I can dig it up.
-Original Message-
From: asterisk-users-boun...@lists.dig
After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a
Segmentation fault.
[root@localhost asterisk-11.1.2]# asterisk -vvc
Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show
I am seeing this as well.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roy Abshire
Sent: Monday, January 07, 2013 1:22 PM
To: Asterisk Users
Subject: [asterisk-users] Outoing Calls Motif Google Voice Calls R
ADTRAN has some interesting Voice Quality Monitoring built into their switches,
routers, etc: http://adtran.com/web/url/vqm
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan at WPF
Sent: Wednesday, June 20, 2012 2:05 PM
To: Asteri
Block them. They are one of the Internet's top bad IP addresses.
http://www.threatstop.com/checkip
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Tuesday, March 06, 2012 7:29 PM
To: asteris
Snom is an OEM of the Konftel.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of brya...@zktech.com
Sent: Sunday, January 08, 2012 12:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Some ideas:
* http://www.clearone.com/voip-conference-phones.html
* http://www.konftel.com/Products/Konftel300IP
*
http://www.polycom.com/products/voice/conferencing_solutions/conference_phones/soundstation/soundstation_duo.html
We have tested all of these in our lab but I would prefer not to be
: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hint'ing with XMPP?
2011/12/5 Jamie A. Stapleton
mailto:jstaple...@computer-business.com>>
I have not ever done what you are talking about.
However, I can tell you that our Openfire XMPP server
I have not ever done what you are talking about.
However, I can tell you that our Openfire XMPP server has similar functionality
because of their Asterisk-IM Plugin.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jay R. Worthington
Se
Some ideas:
* http://www.clearone.com/voip-conference-phones.html
* http://www.konftel.com/Products/Konftel300IP
*
http://www.polycom.com/products/voice/conferencing_solutions/conference_phones/soundstation/soundstation_duo.html
We have tested all of these in our lab but I would prefer not to be
exten => accou...@gmail.com,1,Answer()
exten => accou...@gmail.com,n,Wait(2)
exten => accou...@gmail.com,n,SendDTMF(1)
exten => accou...@gmail.com,n,Dial(SIP/device1)
exten => accou...@gmail.com,1,Answer()
exten => accou...@gmail.com,n,Wait(2)
exten => accou...@gmail.com,n,SendDTMF(1)
exten => ac
Many providers do not allow for caller ID name to be sent.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Friday, June 10, 2011 5:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-
: [asterisk-users] SIP/IAX guest access?
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
On 06/09/2011 08:50 PM, Jamie A. Stapleton wrote:
> Guest calls go to the context specified in [general] of sip.conf.
Thx. Is this valid for IAX2 also?
- -S
- --
(o_ Stefan Gofferje| S
Guest calls go to the context specified in [general] of sip.conf.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan Gofferje
Sent: Thursday, June 09, 2011 1:40 PM
To: Asterisk Users Mailing List - Non-Comm
: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cannot call to my server with SIP
Op 22-04-11 23:49, Jamie A. Stapleton schreef:
> I can see your server just fine...
>
> -bash-3.2# ./svmap.py xen8.vandervlis.nl
> | SIP Device | User Agent |
I can see your server just fine...
-bash-3.2# ./svmap.py xen8.vandervlis.nl
| SIP Device | User Agent | Fingerprint |
--
| 91.198.178.28:5060 | Asterisk PBX 1.6.2.9-2+squeeze1 | disabled|
However,
No problem. You just specify accountn...@gmail.com.
exten => accountn...@gmail.com,1,Answer()
exten => accountn...@gmail.com,n,Wait(2)
exten => accountn...@gmail.com,n,SendDTMF(1)
exten => accountn...@gmail.com,n,Dial(SIP/devicename)
From: asterisk-users-boun...@lists.digium.com
[mailto:asteris
Have you read page 312 of Asterisk: The Future of Telephony
(http://cdn.oreilly.com/books/9780596510480.pdf)?
"there are a few things that need to be
added in order to get it to function. First off, Asterisk needs to have an IMAP
client
installed so that it can communicate with the IMAP server.
iptables -L -v
will give you the IP address that was banned
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Darnell
Sent: Monday, March 07, 2011 1:50 PM
To: Asterisk Users Mailing List - Non-Commercial Di
We have had very good results with nexVortex.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent A. Torrenga
Sent: Thursday, March 03, 2011 11:22 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP Provider Recommenda
http://sipera.com/ is one such product.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan Hisham
Sent: Monday, February 28, 2011 9:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] aster
Just add something like this to your dialplan:
exten=>1234,1,Dial(SIP/u...@domain.com)
Then, when you dial 1234 on your XLite, it will connect you to u...@domain.com.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Beh
1. Per http://www.voip-info.org/wiki/view/Asterisk+video: Asterisk does not
provide any video transcoding capabilities
2. You can turn off video support on a peer like this:
disallow=h261
disallow=h263
disallow=h263p
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun.
We use DISA (http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA) to access
our entire [features] context from our cell phones.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Regal
Sent: Friday, November 05, 2010 11:11 AM
To: ast
Only 100? We had a single server over 300.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria
Sent: Saturday, October 30, 2010 9:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
nexVortex (http://bit.ly/9bEw9e) can do this. They use Global for TF. They
can support both US and CA origination.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joe Freeman
Sent: Tuesday, September 14, 201
Have a Cisco 7975g running SIP firmware version 8.3.4. Many things are broken
with Asterisk.
1) BLF doesn't work
2) MWI doesn't work
3) Sometimes the calls get "stuck" on the display
4) Sometimes MOH works
5) Headset jack doesn't work
Can anyone recommend a version of the SIP firmware for the C
Anyone know of an ITSP that can offer DDIs (or DIDs) from India?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http:
Might be worth your time to check out: http://www.humbuglabs.org/
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Zulu
Sent: Saturday, August 07, 2010 3:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
A packet capture would be most useful. Then, you could review your SDP with
your provider.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy Beak
Sent: Friday, July 23, 2010 7:27 AM
To: Asterisk Users Mail
Have you tried setting
externip=
In the [general] of your sip.conf?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal Goltzman
Sent: Monday, July 05, 2010 1:58 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject
The IP 69.175.35.186 has just been banned by Fail2Ban after 293 attempts
against our server.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Timms
Sent: Thursday, July 01, 2010 11:32 AM
To: Asterisk Users Mailing List - Non-Commer
We are running Asterisk 1.6.2.7-rc1 and SfA without problem. What version are
you running?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner
Sent: Tuesday, April 27, 2010 9:54 AM
To: asterisk-use
us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Jamie A. Stapleton
CBSi - Connecting your problems with solutions.
Telephone: (
s] Which H.323 to use in Ast 1.6
Could you share your config for the Asterisk and Avaya side too? Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie A.
Stapleton
Sent: Wednesday, February 24, 2010
I have always used ooh323 between Avaya and Asterisk.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Tuesday, February 23, 2010 2:24 PM
To: 'Asterisk Users List'
Subject: [asterisk-users] Which H.323 to use in Ast
Have you considered using IAX instead of SIP? IAX2 is a VoIP protocol that
carries both signaling and media on the same port:
http://en.wikipedia.org/wiki/Inter-Asterisk_eXchange
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mosbah.
n Wed, Jan 27, 2010 at 8:32 PM, Jamie A. Stapleton
wrote:
> In this case, a SIP provider would not be required.
>
> Obviously, you will need ports on your EPBX to connect the Digium card to.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [
In this case, a SIP provider would not be required.
Obviously, you will need ports on your EPBX to connect the Digium card to.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Siju George
Sent: Wednesday, Janua
Could use the free http://www.sipgate.com/one for some testing (assuming that
Asterisk is connected to the Internet)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UIT DEVELOPMENT
Sent: Tuesday, January 05, 2
:
http://lists.digium.com/mailman/listinfo/asterisk-users
Jamie A. Stapleton
CBSi - Connecting your problems with solutions.
Telephone: (804) 412-1601
Facsimile: (804) 412-1611
VideoConf: callto:jstapleton.computer-business.com
Meet me on LinkedIn<http://www.linkedin.com/in/jstapleton>
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