to the gateway
(in this case asterisk)
If you still have problems I may be able to dig up some configs for you??
Cheers
Jason
Jason Penton
PhD Candidate
Department of computer Science
Rhodes University
Tel: +27 46 603 8640
Mobile: +27 82 376 6811
VoIP: sip:[EMAIL PROTECTED]
Email: [EMAIL PROTECTED
How does IAX authenticated transfer work? Is there
any documentation available? Mark spoke about it in the paper comparing SIP and
IAX. However I cant seem to find additional info on it
Jason
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Well Andres is right but there are numerous problems with quite a few SIP
clients that do NOT follow the the SIP RFC correctly. There is a problem
with dialog creation in a number of SIP products out there. SIP dialog
creation is the critical part of the spec that supports parallel forking -
so be
Try using $AGI-stream_file(filename)
There are built-in AGI commands - you don't have to use exec for all
commands.
Hope this helps
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Bruce Marler
Sent: 07 July 2004 08:43 AM
To: [EMAIL
Hi All
I have a strange problem using IAX2. When placing a call to my IAX clients
(firefly) via the Asterisk dialplan all works great. However trying to
initiate a call via the manager interface to the IAX client using the
following command results in an error:
Action: Originate
Channel:
no compatible codecs
check under your network settings that you have all the
codecs selected
and obviously type IAX
Jason Penton wrote:
Hi All
I have a strange problem using IAX2. When placing a call to
my IAX clients
(firefly) via the Asterisk dialplan all works great.
However trying
is your friend, looks at the capibilities asterisk
is sending
in it's NEW message
Jason Penton wrote:
Hi Adam
Done all that but still the same problem.
Do you have any other ideas?
Cheers
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi All
Does anyone know how I can get more information about an incoming SIP call
from a SIP proxy. Like FWD or any other SER proxy. My * box shows the
channel name as:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Adam Goryachev
Sent: 26 May
Hi all
Does nayone know how to get more specific info about an incoming SIP call
from a SIP proxy like FWD or any other SER proxy. All incmoing calls into my
* box from FWD and other SER proxies have the following channel name:
SIP/-081833b8 or something similar but with the same format (random)
Hi all
Does anyone have any experience with an E1 Channel bank connected to
Asterisk. I know form an earlier post that Orion telecom and Valiant Telecom
make them. But does anyone know if/how well they work with Asterisk.
Thanks in advance
Jason
-Original Message-
From: [EMAIL
Hi all
Does anyone have any experience with an E1 Channel bank connected to
Asterisk. I know form an earlier post that Orion telecom and Valiant Telecom
make them. But does anyone know if/how well they work with Asterisk.
Thanks in advance
Jason
___
Hi all
Does anyone know where I can get hold of the German 1TR6 ISDN signalling
protocol specification.
Thanks
Jason
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Subject: Re: [Asterisk-Users] ISDN
Jason Penton wrote:
Hi all
Does anyone know where I can get hold of the German 1TR6 ISDN
signalling protocol specification.
Thanks
Jason
Is that still used? I thought they were 100% CTR4 these days.
Regards,
Steve
Hi All
I have noticed a problem with dtmf reception on asterisk's side from H.323
clients (specifically clients sending in-band dtmf like NM). Asterisk v.
0.5.0 works perfectly while the latest release (0.7.1) never works. I am
going ot look at the code later to see what has been changed
Anybody
] RE: Fax
Hi,
On Tue, 13 Jan 2004 at 21:06, Jason Penton wrote:
(I have successfully managed to receive faxes thru my isdn
card so I
don't see why I shouldn't be able to send them).
that's interesting, as in my tests it was just the other way around.
I can send faxes through my AVM
Hi All
I have just a quick question regarding app_txfax for Asterisk.
When I send a fax from asterisk to a traditional fax machine connected to
asterisk via the digium analog card everything works perfectly. However the
same fax machine on the public telephoine network results in errors (looks
, Jason Penton wrote:
I have just a quick question regarding app_txfax for Asterisk.
When I send a fax from asterisk to a traditional fax machine
connected to asterisk via the digium analog card everything works
perfectly. However the same fax machine on the public telephoine
network results
I would love to try it Thanks very
much
Jason
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Azher Amin
Sent: 02 December 2003 07:03 AM
To:
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
PREPAID APPLECATION
Hi Bart,
I would like to test
Asterisk is an H.323 gateway - if you want it to be ;-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David J
Carter
Sent: 06 November 2003 07:20 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] H323 Gateway
Hi all,
Anyone know of a small H323
Hi Staffan
Yeah I managed to get everything setup quite nicely with SER and
Asterisk on the same box. I am however still doing testing so I may
still find some problems - but for the time being all is ok.
What I did was to bind two real-world ip addresses to the same NIC on my
box. I then
Hi All
I have come across a really strange bug using app_festival. This bug
however only occurs when I am calling from a sip client. When calling
from an H.323 client everythign works fine.
After initiation of festival within the extensions file - there seems to
be an error coming from rtp.c
The gazel 128 PCI cards are great - never had any problems. They also
offer a USB one that is supported by ISDN4Linux. You can get these
products from www.bewan.com
HTH
Cheers
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Marc V.
Liotier
I have asterisk and SER running on the same machine perfectly. As far as
I am concerned the best way to do it is to have two ip addresses for the
same ethernet interface. That way you can bind asterisk to one IP
address and SER to the other. This way you don't have to use
non-standard ports for
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