_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jigar Joshi
Sent: Tuesday, October 19, 2010 1:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to connect asterisk PBX to PSTN
On 10-10-18 11:01 PM, Barry Miller wrote:
On Mon, Oct 18, 2010 at 07:58:07PM -0400, Asterisk Development Team wrote:
On 10-10-18 07:54 PM, Asterisk Development Team wrote:
For a full list of changes in the current release candidate, please see the
ChangeLog:
This would be overkill, but I thought I would mention these products in case
someone was looking for a more robust/industrial automation solution that
could be integrated with the linux command line and asterisk. I do use
asterisk as a remote control device for a few non-critical inputs (air
How did the setup work as far as extensions on the Inter-Tel system
contacting extensions on the asterisk system?
On Thu, Oct 14, 2010 at 9:56 AM, Justin Sherrill
justin.sherr...@americanrocksalt.com wrote:
We have it integrated, but differently; we have 2 T1 voice lines, and a
4-port
Hello,
I'm trying to send a tif file, using Fax for Asterisk and the call is
executed, but when I get the reINVITE with T.38 data, the local server
doesn't recognize that we have this capability and sends a 488 message.
These are the logs:
--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---
Hi Cassius
On the iax conf for both box change the secret parameter with remotesecret.
This is a undocumented change between Asterisk 1.6.1.X and Asterisk 1.6.2.X
Regards
2010/10/18 Cassius Smith cass...@cassius.org
I'm having trouble getting an IAX2 connection between a couple of
servers.
Greeting list,
I hope this isn't a lazy question. I have been running
TDM400P and TDM410P cards in Dell PowerEdge Servers for a few years now. We
are moving from physical servers to VMWARE servers. What opportunities
should I expect moving these cards into the new machines?
On Tue, Oct 19, 2010 at 10:36 AM, VoIP Question voip.quest...@gmail.com wrote:
Hello,
I'm trying to send a tif file, using Fax for Asterisk and the call is
executed, but when I get the reINVITE with T.38 data, the local server
doesn't recognize that we have this capability and sends a 488
Hi,
I think as more than one already replied this is feasible, but I would like
to underline that it's better to have a look to that book at least to know
how to ask and provide more details to this mailing list by using a more
VoIP/Asterisk standard language.
BTW I think you would like to
It's set to yes for this peer.
also t38pt_udptl is set to yes.
:(
On Tue, Oct 19, 2010 at 5:12 PM, David Backeberg dbackeb...@gmail.comwrote:
On Tue, Oct 19, 2010 at 10:36 AM, VoIP Question voip.quest...@gmail.com
wrote:
Hello,
I'm trying to send a tif file, using Fax for Asterisk
Thanks All,
I will look into book for sure.
On Tue, Oct 19, 2010 at 8:50 PM, Marino Punturieri map...@gmail.com wrote:
Hi,
I think as more than one already replied this is feasible, but I would like
to underline that it's better to have a look to that book at least to know
how to ask and
On Tue, Oct 19, 2010 at 11:21 AM, VoIP Question voip.quest...@gmail.com wrote:
It's set to yes for this peer.
also t38pt_udptl is set to yes.
:(
You don't say anything about what you're trying to send / receive against.
Here's how you should troubleshoot:
* start with a 'real fax machine'
On Tue, Oct 19, 2010 at 10:23 AM, marvin horst fivehor...@gmail.com wrote:
How did the setup work as far as extensions on the Inter-Tel system
contacting extensions on the asterisk system?
It worked, I dare say, flawlessly. Well, as flawlessly as Inter-Tel
worked. Still had to watch out for
We don't have an ATA and fax machine.
The whole point (as I specified in the header and initial message) is the
attempt to use Fax for Asterisk to send the message.
As I showed in the logs, the remote carrier sends a proper T.38 reINVITE,
but our Asterisk doesn't accept, despite the fact that
From what I have read over the last few months, you should invest in Motrin
before trying T.38 faxing with or without FFA - it can (possibly) be done,
but it has beaten some folks into the ground trying it.
Could be a codec issue.
--
On Tue, Oct 19, 2010 at 10:44 AM, Andrea Sannucci asannu...@gmail.com wrote:
On the iax conf for both box change the secret parameter with remotesecret.
This is a undocumented change between Asterisk 1.6.1.X and Asterisk 1.6.2.X
This is incorrect, chan_iax.so does not have such a parameter.
On Mon, 18 Oct 2010 13:09:50 +0200, Gilles codecompl...@free.fr
wrote:
I'm sure someone has already tried this: I use a couple of electric
heaters to heat my office.
Thanks everyone for the great feedback. Following Steve Edward's
advice, I won't automate the process and will only switch the
Hi Paul,
I spent two days to conect two Asterisk BOX (1.6.2.13) with IAX with
username and password.
Only when i changed secret with remotesecret the connection work.
Maybe you can try the same configuration to confirm this behaviour
Regards
- Bakko
--
On Tue, Oct 19, 2010 at 11:48 AM, VoIP Question voip.quest...@gmail.com wrote:
The whole point (as I specified in the header and initial message) is the
attempt to use Fax for Asterisk to send the message.
Asterisk can handle audio passthrough faxing. I'm talking audio faxing
over SIP. You
I spent two days to conect two Asterisk BOX (1.6.2.13) with IAX with
username and password.
Only when i changed secret with remotesecret the connection work.
I would enable iax debugs and confirm you calls you being
authenticated, and not using a guest account. As I mentioned,
'remotesecret'
On Mon, Oct 18, 2010 at 7:41 PM, Cassius Smith cass...@cassius.org wrote:
Any hints for me?
Server Ottawa (192.168.1.190)
register = Ottawa:ottawaisc...@192.168.1.196
[Toronto]
type=peer
host=dynamic
username=Toronto
secret=TorontoIsFine
Server Toronto (192.168.1.196)
register =
Digium claims that their FFA is the best and most compatible solution and
they give one channel for free, but do not provide support for those that do
not buy more channels, but why buy more channels if the free/test one
doesn't work?
I know they read (and sometimes respond) to this list, so I
I think the generic throw away gmail address will keep many people
from answering...
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux
On Tue, Oct 19, 2010 at 1:01 PM, VoIP Question voip.quest...@gmail.com wrote:
Digium claims that their FFA is the best and most compatible solution and
they give one channel for free, but do not provide support for those that do
not buy more channels, but why buy more channels if the free/test
Hi Cassius,
it may be slightly offsubject but i did connect two 1.6 asterisk boxes, and
the only issue i had is these two statements missing:
calltokenoptional=209.16.236.73/255.255.255.0
requirecalltoken=no
hope it helps!
On Tue, Oct 19, 2010 at 12:54 PM, Paul Belanger
On 10-10-19 10:46 AM, Danny Nicholas wrote:
Greeting list,
I hope this isn’t a “lazy” question. I have been running TDM400P and
TDM410P cards in Dell PowerEdge Servers for a few years now. We are
moving from physical servers to VMWARE servers. What opportunities
should I expect moving these
Dear, I have an Asterisk PBX with two E1 cards: Digium TE122 and Sangoma
A101D. Sangoma card has SNMP support but Digium card has not, and also SNMP
does't give me ral time information.
Within CLI Asterisk I execute dahdi show channels but I don't get
information about channels usage.
What is
- Alejandro Cabrera Obed aco1...@gmail.com wrote:
Dear, I have an Asterisk PBX with two E1 cards: Digium TE122 and Sangoma
A101D. Sangoma card has SNMP support but Digium card has not, and also SNMP
does't give me ral time information.
Within CLI Asterisk I execute dahdi show
Hi Paul,
maybe there is some think wrong on iax.
if I set remotesecret on IAX2 extension the call from Server A to Server B
work but not authenticated and host is set to dynamic (normaly if is a IP
authentication on host parametre I put the IP)
If I set secret on two box and both are
On 10/19/2010 10:48 AM, VoIP Question wrote:
We don't have an ATA and fax machine.
The whole point (as I specified in the header and initial message) is
the attempt to use Fax for Asterisk to send the message.
As I showed in the logs, the remote carrier sends a proper T.38
reINVITE, but
On 10/19/2010 12:01 PM, VoIP Question wrote:
Digium claims that their FFA is the best and most compatible solution
and they give one channel for free, but do not provide support for those
that do not buy more channels, but why buy more channels if the
free/test one doesn't work?
I know they
Fair enough Kevin :-) It's just that your documentation for this product is
so limited that without extensive search online and the assistance of
others, it would have been impossible for us to make any progress and we
haven't reached the ReceiveFax part yet ;)
Anyway, specifically, we installed
From: marvin horst [mailto:fivehor...@gmail.com]
Sent: Tuesday, October 19, 2010 10:23 AM
To: Justin Sherrill; asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] integrate Intertel Axxess with Asterisk
How did the setup work as far as extensions on the Inter-Tel system
Hi listers!
Have a problem with distortion in some analog lines. When some call comes in
from PSTN the sound is really distorte, nothing can be understanded, but
Internal calls work ok.
Funny thing is that when I start/stop asterisk,dahdi, and wanrouter services
eveything goes fine again. This
On Tue, Oct 19, 2010 at 09:59:34AM -0400, Leif Madsen wrote:
On 10-10-18 11:01 PM, Barry Miller wrote:
On Mon, Oct 18, 2010 at 07:58:07PM -0400, Asterisk Development Team wrote:
On 10-10-18 07:54 PM, Asterisk Development Team wrote:
For a full list of changes in the current release
Hi
We are facing a problem for orphaned parked calls, we have the following
config:
asterisk -1.4.22.1
dahdi-linux-complete-2.2.0.2+2.2.0
and when we get an incoming call and after it gets parked, after some set
time (here its 2 min), it goes back to the operator, but the problem is that
Hi ,
Please, I am trying to understand the hardware installation on asterisk and I
have some doubt. If I uncomment the hardware type in /etc/dahdi/modules and
then I run the dahdi_genconf , It create the dahdi_channels and system.conf.
Therefore, it is created with a kind of signalling
On 10/19/10 10:03 PM, Flavio Miranda wrote:
Please, I am trying to understand the hardware installation on asterisk
and I have some doubt. If I uncomment the hardware type in
/etc/dahdi/modules and then I run the dahdi_genconf , It create the
dahdi_channels and system.conf.
Therefore, it is
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
Just one more question, what it means the RED under alarms when I type dahdi
show status. It should be OK?
Thanks for your guidance!
Date: Tue, 19 Oct 2010 22:38:25 -0500
From: sruff...@digium.com
To:
39 matches
Mail list logo
