On 01/08/2013, at 2:20 PM, Zoltán Fekete bl...@gyoz.info wrote:
2013/8/1 Joshua Colp jc...@digium.com
Larry Moore wrote:
On 31/07/2013 8:08 PM, Joshua Colp wrote:
Zoltán Fekete wrote:
Thank You Larry!
I have discussed with my provider. They are not able to insert the
T38MaxBitRate
2013/8/1 Joshua Colp jc...@digium.com
Larry Moore wrote:
On 31/07/2013 8:08 PM, Joshua Colp wrote:
Zoltán Fekete wrote:
Thank You Larry!
I have discussed with my provider. They are not able to insert the
T38MaxBitRate value into the sip answer. :(
Thank You Larry!
I have discussed with my provider. They are not able to insert the
T38MaxBitRate value into the sip answer. :(
https://gist.github.com/anonymous/6120148 (line 559)
That means we are not able to passtrough T38 Faxes with any asterisk
version at all?
What do you mean? Am I able to
Zoltán Fekete wrote:
Thank You Larry!
I have discussed with my provider. They are not able to insert the
T38MaxBitRate value into the sip answer. :(
https://gist.github.com/anonymous/6120148 (line 559)
That means we are not able to passtrough T38 Faxes with any asterisk
version at all?
What do
On 31/07/2013 8:08 PM, Joshua Colp wrote:
Zoltán Fekete wrote:
Thank You Larry!
I have discussed with my provider. They are not able to insert the
T38MaxBitRate value into the sip answer. :(
https://gist.github.com/anonymous/6120148 (line 559)
That means we are not able to passtrough T38
Larry Moore wrote:
On 31/07/2013 8:08 PM, Joshua Colp wrote:
Zoltán Fekete wrote:
Thank You Larry!
I have discussed with my provider. They are not able to insert the
T38MaxBitRate value into the sip answer. :(
https://gist.github.com/anonymous/6120148 (line 559)
That means we are not able to
On 22/07/2013 5:40 AM, Zoltán Fekete wrote:
Hi!
I have exactly the same problem on asterisk 1.8.22.0 and also on
separate 11.2.1 when sending fax to PSTN.
Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper softphone.
SpanDsp also works without any problem on my box.
As I
if that parameter is missing, then the code would in fact default to 2400
as a safe value.
Kevin Larsen - Systems Analyst
From: Zoltán Fekete bl...@gyoz.info
To: asterisk-users@lists.digium.com,
Date: 07/21/2013 04:40 PM
Subject:[asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through
On 23/07/2013 6:18 AM, Kevin Larsen wrote:
The a=T38MaxBitRate issue you refer to was one that was actually
discovered at my company and submitted by a colleague. It was fixed in
11.3.0 and 1.8.21.0. However, I think that it wouldn't help based on the
description below being that the parameter
On 22/07/2013 10:19 PM, Larry Moore wrote:
On 22/07/2013 5:40 AM, Zoltán Fekete wrote:
Hi!
I have exactly the same problem on asterisk 1.8.22.0 and also on
separate 11.2.1 when sending fax to PSTN.
Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper
softphone.
SpanDsp also works
Hi!
I have exactly the same problem on asterisk 1.8.22.0 and also on separate
11.2.1 when sending fax to PSTN.
Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper softphone.
SpanDsp also works without any problem on my box.
As I remember it was a bug in 1.8.1.x that the
can anyone help me with this case?
thank you,
Leo
Hi there,
I setup realtime asterisk 10.3.0 with backend mysql server. everything seems to
work fine except when I tried to enable the extensions for dialplan to be
obtained from mysql, I got an empty dialplan. I am not sure why this
Hello,
is there anyone that can give me some more information on these
deadlocks ?!
How can these deadlocks occur and what is good practise to avoid these
problems ??
Jonas.
Original Message
Subject:Re: [asterisk-users] Asterisk CLI unresponsive
Date: Fri,
On 6 February 2012 10:45, Jonas Kellens jonas.kell...@telenet.be wrote:
**
Hello,
is there anyone that can give me some more information on these
deadlocks ?!
How can these deadlocks occur and what is good practise to avoid these
problems ??
Jonas.
The only way to avoid deadlocks is
On 02/06/2012 12:14 PM, Steve Davies wrote:
On 6 February 2012 10:45, Jonas Kellens jonas.kell...@telenet.be
mailto:jonas.kell...@telenet.be wrote:
Hello,
is there anyone that can give me some more information on these
deadlocks ?!
How can these deadlocks occur and what is
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive
--
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On 02/06/2012 12:25 PM, isr...@gmail.com wrote:
Your running into a bug and the only way to solve it is to report it and debug
it and hope for a fix
There is no way someone can help without it being debugged and knowing what's
causing it to lockup
The only key to unlcock it when it gets
On 12-02-06 09:15 AM, Jonas Kellens wrote:
On 02/06/2012 12:25 PM, isr...@gmail.com wrote:
Your running into a bug and the only way to solve it is to report it
and debug it and hope for a fix
There is no way someone can help without it being debugged and knowing
what's causing it to lockup
The
On 02/06/2012 03:19 PM, Paul Belanger wrote:
On 12-02-06 09:15 AM, Jonas Kellens wrote:
On 02/06/2012 12:25 PM, isr...@gmail.com wrote:
Your running into a bug and the only way to solve it is to report it
and debug it and hope for a fix
There is no way someone can help without it being
, 2012 9:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive
On 02/06/2012 03:19 PM, Paul Belanger wrote:
On 12-02-06 09:15 AM, Jonas Kellens wrote:
On 02/06/2012 12:25 PM, isr
On 12-02-06 09:23 AM, Jonas Kellens wrote:
On 02/06/2012 03:19 PM, Paul Belanger wrote:
On 12-02-06 09:15 AM, Jonas Kellens wrote:
On 02/06/2012 12:25 PM, isr...@gmail.com wrote:
Your running into a bug and the only way to solve it is to report it
and debug it and hope for a fix
There is no
- Original Message -
From: Jonas Kellens jonas.kell...@telenet.be
To: isr...@gmail.com, Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Monday, February 6, 2012 8:15:50 AM
Subject: Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive
-Scanned-By: MIMEDefang 2.67 on 205.211.164.50
Subject: [asterisk-users] Fwd: Re: Asterisk as a Condo door
opener/intercom
X-BeenThere: asterisk-users@lists.digium.com
X-Mailman-Version: 2.1.11
Precedence: list
Reply-To: dbc_aster...@advan.ca,
Asterisk Users Mailing List - Non-Commercial
Asterisk as a phone system makes perfect sense in a condo. You can get
all the DID's you want and eliminate costs for the owners. You can offer
standard FXO for people who don't care and IP sets for people who want
to upgrade to feature sets.
Your door openner is a piece of cake.
1. Create an
Juan.
Linux User #441131
-- Forwarded message --
From: Juan David Diaz juanch...@gmail.com
Date: Tue, Nov 16, 2010 at 1:38 PM
Subject: HA - asterisk service is not starting
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Hi Asterisk
On 16 November 2010 22:43, Juan David Diaz juanch...@gmail.com wrote:
Juan.
Linux User #441131
Maybe best on the linux-ha lists...
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New to
David @ULC schrieb:
If I use below code in my sip.conf ,
[123]
type=peer
qualify=no
port=5060
nat=no
insecure=very this is very important
host=voiper.ipkall.com
dtmfmode=rfc2833
context=from-pstn
canreinvite=no
how will call understand that where I have to land as we DO
this is like the bible
http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.com/books/9780596510480.pdf
2009/1/14 Philipp Kempgen philipp.kemp...@amooma.de
David @ULC schrieb:
If I use below code in my sip.conf ,
[123]
type=peer
qualify=no
port=5060
nat=no
I have an account with FWD and I have configured my SIP.conf with
[fwd]
type=friend
secret=password
username=901835
host=fwd.pulver.com
But when I am trying to dial out my own DID , I dont see any call landing in
asterisk.
In extension.conf (vicidial) file I have
exten = 2062036895
I also tried but cant see any call landing up in asterisk.
Btw, how to find out whether a call is landing in Asterisk or not ?
[123]
type=peer
qualify=no
port=5060
nat=no
insecure=very this is very important
host=voiper.ipkall.com
dtmfmode=rfc2833
context=from-pstn
canreinvite=no
ngrep port 5060
or tcpdum port 5060
By default asterisk runs on port 5060, that way you can see if your getting
the signal or not.
Jai Rangi
Buy SIP DID www.didforsale.com
free Trial now purchase required
On Tue, Jan 13, 2009 at 1:13 PM, David @ULC ucoms2...@gmail.com wrote:
I also tried but
[r...@vicidialnow ~]# ngrep port 5060
-bash: ngrep: command not found
[r...@vicidialnow ~]# tcpdum port 5060
-bash: tcpdum: command not found
[r...@vicidialnow ~]#
Also, is my SIP configuration is correct ?
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Sorry for the typo,
tcpdump port 5060
ngrep you can download the rpm (google) easy to install
http://rpm.pbone.net/index.php3/stat/4/idpl/1127130/com/ngrep-1.38-1.i386.rpm.html
rpm -ivh
[123]
type=peer
qualify=no
port=5060
nat=no
insecure=very this is very important
host=voiper.ipkall.com
dtmfmode=rfc2833
context=from-pstn
canreinvite=no
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David @ULC schrieb:
[r...@vicidialnow ~]# ngrep port 5060
-bash: ngrep: command not found
aptitude install ngrep
[r...@vicidialnow ~]# tcpdum port 5060
-bash: tcpdum: command not found
aptitude install tcpdump
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany -
Anyone using FWD with Asterisk ?
On Wed, Jan 14, 2009 at 2:40 AM, David @ULC ucoms2...@gmail.com wrote:
I have an account with FWD and I have configured my SIP.conf with
[fwd]
type=friend
secret=password
username=901835
host=fwd.pulver.com
But when I am trying to dial out my own DID ,
If I use below code in my sip.conf ,
[123]
type=peer
qualify=no
port=5060
nat=no
insecure=very this is very important
host=voiper.ipkall.com
dtmfmode=rfc2833
context=from-pstn
canreinvite=no
how will call understand that where I have to land as we DO NOT provide our
IP in fwd
When I logged in to my IPKall website ,
I see SIP Proxy: as fwd.pulver.com Do I need to change it to my PUBLIC or
STATIC IP ?
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- Forwarded message from Gilad Ben-Yossef [EMAIL PROTECTED] -
From: Gilad Ben-Yossef [EMAIL PROTECTED]
Organization: Codefidence ltd. A name you can trust.
To: Linux-IL linux-il@linux.org.il
Subject: [Announcement] Asterisk-IL mailing list
X-Bogosity: Unsure [50.0%]
X-listar-version:
Michael Graves wrote:
Sounds like reinvite troubles. Once the SIP endpoints are both in the
call the server (FWD) will get out of the way allowing the two SIP
clients to connect directly. There can be cases where you can connect
through the server but not directly, usually because of NAT traversal
I asked my friend to setup FWD and call me to my *
However, it did not matter which codec we used, after three seconds the
connection was cut.
Why? and how to make it stabled?
bye
Ronald
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Asterisk-Users@lists.digium.com
Sounds like reinvite troubles. Once the SIP endpoints are both in the
call the server (FWD) will get out of the way allowing the two SIP
clients to connect directly. There can be cases where you can connect
through the server but not directly, usually because of NAT traversal
failure at one end or
.
-Original Message-
From: Geoffrey S. Mendelson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 25, 2005 5:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] [Fwd: Re: [Asterisk-biz] bellster.net -
GREATadvance]
On Tue, Jan 25, 2005 at 02:43:27PM
Shoval Tomer wrote:
As far as I know it's not legal to join bellster in Israel.
It means that you're reselling the minutes you buy from the telco
company.
Wouldn't you need to be selling them to be reselling?
Does that make DISA illegal, and VoIP connections between offices if you
dial out the
On Wed, Jan 26, 2005 at 04:11:04PM -0600, Michael Giagnocavo wrote:
Shoval Tomer wrote:
As far as I know it's not legal to join bellster in Israel.
It means that you're reselling the minutes you buy from the telco
company.
What about the comming real soon now, cable company VOIP
Michael Giagnocavo wrote:
Shoval Tomer wrote:
As far as I know it's not legal to join bellster in Israel.
It means that you're reselling the minutes you buy from the telco
company.
Wouldn't you need to be selling them to be reselling?
Does that make DISA illegal, and VoIP connections between
Message-
From: Geoffrey S. Mendelson [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 27, 2005 12:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] [Fwd: Re: [Asterisk-biz]bellster.net-
GREATadvance]
On Wed, Jan 26, 2005 at 04:11:04PM -0600
Geoffrey S. Mendelson wrote:
On Wed, Jan 26, 2005 at 04:11:04PM -0600, Michael Giagnocavo wrote:
Shoval Tomer wrote:
As far as I know it's not legal to join bellster in Israel.
It means that you're reselling the minutes you buy from the telco
company.
What about the comming real soon now, cable
-Original Message-
Bummer. Glad I don't run a business in Israel. Thought it was bad here
in New Zealand! I'd hate to have my business phone cut off because
someone saw an increased call volume!
It's such a big deal (For instance, here in Guatemala), they have dedicated
people who
Steven P. Donegan wrote:
I don't want to be negative here, but I do believe people who go to do this know the potential risks they face. In many countries (4 of which I have direct, although several year old experience with - all in Asia) taking a local phone line and attaching asterisk to it and
Funny, the only thing I addressed was the direct threat of busting the
contract/acceptable use policy of your Telco/local government. I didn't
go anywhere near the other risks:
1) you mess up your extensions.conf and some bozo - on purpose or
otherwise - runs up some insane bill on that nice
Duane wrote:
I was discussing bellster with a friend of mine, and he made another
point about this service...
I can't imagine how unsettling it would be for my girlfriend to pick up the
phone and hear somebody else on the line. The first time that happened, that'd
be the end of me sharing the
On Tue, Jan 25, 2005 at 02:43:27PM +1100, Duane wrote:
Another small point is that a lot of countries don't have flat rate
calls, and I highly doubt anyone in those countries would be offering
their land lines for this kind of service either. It costs me between 20
and 30c per call to make
, January 24, 2005 10:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Commercial
and Business-Oriented Asterisk Discussion
Subject: Re: [Asterisk-Users] [Fwd: Re: [Asterisk-biz] bellster.net -
GREATadvance]
Duane wrote:
I was discussing bellster with a friend of mine, and he made
On Mon, 2005-01-10 at 08:01 -0600, [EMAIL PROTECTED] wrote:
I am new to asterisk but learn a lot about it to this mailing list and
wiki currently i am facing problem about sip phone i have PA 1688
chipset ip-phone and i have iptel.org sip account i registered locally
and
Hello,
In my sip.conf I have:
;Register and forward FWD numbers to internal extensions
register = FWDNUMBER:[EMAIL PROTECTED]/9500
Which should register Asterisk at FWD and then when any calls are made to
FWDNUMBER those calls should be forwarded to extension 9500 as specified in
the
The codec issues with different services and sip phone are the most
complicated and trusting experience when using Voip services.
I had been able to connect to FWD behind a firewall by using Iaxtel
using g729. Just recently, about a week, every time I tried to call FWD,
the connection simply
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