On 22/07/2011 5:43 AM, Israel Gottlieb wrote:
On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming
kpflem...@digium.com mailto:kpflem...@digium.com wrote:
On 07/21/2011 04:34 PM, Joaquin Sosa wrote:
On Mon, Jul 18, 2011 at 07:58, Steve
Daviesdavies...@gmail.com
On Mon, Jul 18, 2011 at 07:58, Steve Davies davies...@gmail.com wrote:
The magic sauce that you need is T.38 - Asterisk 1.6 supports this
to a limited degree, and your ITSP will need to support it.
The sip.conf.sample file and the voip-info wiki has all the
information you need to try it out.
On 07/21/2011 04:34 PM, Joaquin Sosa wrote:
On Mon, Jul 18, 2011 at 07:58, Steve Daviesdavies...@gmail.com wrote:
The magic sauce that you need is T.38 - Asterisk 1.6 supports this
to a limited degree, and your ITSP will need to support it.
The sip.conf.sample file and the voip-info wiki has
On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming kpflem...@digium.comwrote:
On 07/21/2011 04:34 PM, Joaquin Sosa wrote:
On Mon, Jul 18, 2011 at 07:58, Steve Daviesdavies...@gmail.com wrote:
The magic sauce that you need is T.38 - Asterisk 1.6 supports this
to a limited degree, and your
On 07/21/2011 04:43 PM, Israel Gottlieb wrote:
On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
On 07/21/2011 04:34 PM, Joaquin Sosa wrote:
On Mon, Jul 18, 2011 at 07:58, Steve Daviesdavies...@gmail.com
On Thu, Jul 21, 2011 at 17:39, Kevin P. Fleming kpflem...@digium.com wrote:
We do this in our testing all the time, and it works fine. Since you didn't
specify any particular version of Asterisk, there's no way to associate your
It won't work statement with anything in particular. Given the
On Fri, Jul 22, 2011 at 12:50 AM, Kevin P. Fleming kpflem...@digium.comwrote:
On 07/21/2011 04:43 PM, Israel Gottlieb wrote:
On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
On 07/21/2011 04:34 PM, Joaquin Sosa wrote:
On 11-07-21 05:52 PM, Joaquin Sosa wrote:
On Thu, Jul 21, 2011 at 17:39, Kevin P. Flemingkpflem...@digium.com wrote:
We do this in our testing all the time, and it works fine. Since you didn't
specify any particular version of Asterisk, there's no way to associate your
It won't work statement
Hello guys
I need some help to do works FAX using SIP, anybody know the secret to
this? Have asterisk 1.6.
Thanks!!
--
Enviado do meu celular
Eduardo Carpes
E-mail: car...@bsd.com.br
www.freebsd.org
--
_
-- Bandwidth and
On 18 July 2011 12:20, Eduardo Carpes car...@bsd.com.br wrote:
Hello guys
I need some help to do works FAX using SIP, anybody know the secret to
this? Have asterisk 1.6.
Thanks!!
--
Enviado do meu celular
Eduardo Carpes
E-mail: car...@bsd.com.br
www.freebsd.org
The magic sauce that you
So Steve
I looked this, but, i didn't understood the difference between enable T.38
and T.38 Gateway, this site ttp://www.voip-info.org/wiki/view/T.38 talk
--Asterisk *1.6* support G.711 and T.38 FAX origination and termination.
T.38 gateway features are still in development. --
I know that
Short answer is: dont use it. For the long answer wait for others to
answer that.
On Mon, Jul 18, 2011 at 7:20 AM, Eduardo Carpes car...@bsd.com.br wrote:
Hello guys
I need some help to do works FAX using SIP, anybody know the secret to
this? Have asterisk 1.6.
Thanks!!
--
Enviado do meu
I resoundingly second that.
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
On Jul 18, 2011, at 11:12 PM, C F shma...@gmail.com wrote:
Short answer is: dont use it.
A DID number was dedicated to receive fax, but i have the problem when
getting fax call,
which call will become a normal phone call and no fax was printed. When
fax is detected,
the fax extension is executed and dial the extension of the HT486 device
(firmware 1.0.5.22).
Somehow sending fax out
A DID number was dedicated to receive fax, but i have the problem when
getting fax call,
which call will become a normal phone call and no fax was printed. When
fax is detected,
the fax extension is executed and dial the extension of the HT486 device
(firmware 1.0.5.22).
Somehow sending
Hello everyone!
I tried to send a fax over SIP with an Asterisk Server in the middle (no
Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway
is external). Whenever I start sending a Fax to a PSTN destination, the
Call gets answered and asterisk tries to build a native bridging:
: Tuesday, November 23, 2004 6:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Fax over SIP Problems (sorry for this topic
...)
Hello everyone!
I tried to send a fax over SIP with an Asterisk Server in the middle (no
Digium Cards, etc. installed
Hello!
Elman Efendiyev wrote:
Are you trying to send fax over T.38?
As far I understand * don't support T.38 event when passing packets
trouth.
No, I'm not trying to send the fax over T.38, I am trying to send it in
the voice path by using the G711 alaw codec. This should work, I
think, but it
Sent: Tuesday, November 23, 2004 8:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Fax over SIP Problems (sorry for this
topic ...)
Hello!
Elman Efendiyev wrote:
Are you trying to send fax over T.38?
As far I understand * don't support T.38 event
At 21:20 16-11-2003 -0600, you wrote:
You will need to check with Cisco to see if the ATA188 has the same issues
with faxing as the ATA186.
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_field_notice09186a0080094af7.shtml
Be advised this item regards the ATA186-L series, and the
You will need to check with Cisco to see if the ATA188 has the same issues
with faxing as the ATA186.
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_field_notice09186a0080094af7.shtml
Dave Weis wrote:
Should I expect a standard fax machine connected to an ata-188 connected
to an
At 20:34 14-11-2003 -0600, you wrote:
Should I expect a standard fax machine connected to an ata-188 connected
to an asterisk server, connected to a pri fed from a cisco 7206vxr to work
correctly? It needs to have a standard fax machine, receiving and emailing
it won't be acceptable.
I have gotten
Am Sam, 2003-11-15 um 11.22 schrieb Florian Overkamp:
At 20:34 14-11-2003 -0600, you wrote:
Should I expect a standard fax machine connected to an ata-188 connected
to an asterisk server, connected to a pri fed from a cisco 7206vxr to work
correctly? It needs to have a standard fax machine,
Should I expect a standard fax machine connected to an ata-188 connected
to an asterisk server, connected to a pri fed from a cisco 7206vxr to work
correctly? It needs to have a standard fax machine, receiving and emailing
it won't be acceptable.
Thanks
dave
--
Dave Weis I
Hello.
Has someone been able to make work faxes over sip, i have one mp108
fxo and one mp108 fxs, my setup is :
telco analog line - mp108fxo - Asterisk -- mp108fxs
--- fax machine
1) Asterisk detects the tone from the sending fax ( i am receiving ) but
looks for
On Thu, 2003-09-04 at 01:18, Ing. Angel Gomez Garcia wrote:
Hello.
Has someone been able to make work faxes over sip, i have one mp108
fxo and one mp108 fxs, my setup is :
telco analog line - mp108fxo - Asterisk -- mp108fxs
--- fax machine
1) Asterisk detects
At 15:30 26-6-2003 -0300, you wrote:
I've tested ATA186 with a cisco827 as the H323 (or SIP) gateway
and I could transmite the fax without problem.
I get erros when sending faxes only when I user asterisk. :~
any tips?
I imagine the Cisco stuff uses T30/T38 amongst
On Thu, 26 Jun 2003 09:01:21 +0200
Florian Overkamp [EMAIL PROTECTED] wrote:
Hi there,
I have made this setup work without any special modifications. I expect it
raises some strict requirements on the latency of your IP network, so that
might be an issue.
I've never tried it with SIP, but I have faxed between to asterisk boxes
on the same network via IAX and IAX2. The secret was to set the codec
to ulaw or alaw. (Certain codecs, such as GSM, compress the data too
much for the fax machines to be able to communicate effectively.)
Jared
On Thu,
Have you tried limiting your fax machines to a lower baud rate like 9600.
I know on Vonage this seems to help.
- Original Message -
From: Eduardo Goncalves [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 26, 2003 10:10 AM
Subject: Re: [Asterisk-Users] Fax and SIP
On Thu
REMOVE
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 26, 2003 4:01 AM
Subject: Re: [Asterisk-Users] Fax and SIP
At 16:01 25-6-2003 -0300, you wrote:
Hi list,
I have the following scenario, and want to know what I
Jim == Jim Flagg [EMAIL PROTECTED] writes:
Jim Have you tried limiting your fax machines to a lower baud rate
Jim like 9600. I know on Vonage this seems to help.
Speaking of which, IIRC the docs for the ata mention that fax at
greater than 9600 is b0rked up to a recent firmware release.
You
On 26 Jun 2003 12:53:40 -0400
James H. Cloos Jr. [EMAIL PROTECTED] wrote:
Jim == Jim Flagg [EMAIL PROTECTED] writes:
Jim Have you tried limiting your fax machines to a lower baud rate
Jim like 9600. I know on Vonage this seems to help.
Speaking of which, IIRC the docs for the ata
Hi list,
I have the following scenario, and want to know what I have to do to transmit
faxes trought this link:
|cisco-ata186|sip-|asterisk|---EM alaw link-PSTN
The codec used is g711a.
When I try to transmit a fax I receive a TX FUNCTION WAS NOT
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