Hello,
There are some parameters in the ISUP data (coming into the network via
SIP-T packets) that need to be translated into SIP headers to be used by
asterisk for proper call routing. What gateways are available to accomplish
this?
Thanks,
Elliot
--
On 05/23/2011 03:26 AM, Elliot Murdock wrote:
There are some parameters in the ISUP data (coming into the network
via SIP-T packets) that need to be translated into SIP headers to
be used by asterisk for proper call routing. What gateways are
available to accomplish this?
What do you mean
Hello,
Thank you for the reply.
Basically, currently, Asterisk does not seem to support SIP-T decoding.
Accordingly, I am looking for a third-party gateway that can bridge (along
with decoding any ISUP data) a server sending SIP-T packets to the Asterisk
server. The topography would be as
Anyone know how I can get SIP T working w/ Asterisk?
TIA,
Jon
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5 apr 2006 kl. 16.40 skrev Jon Weisman:
Anyone know how I can get SIP T working w/ Asterisk?
Start with explaining your definition of SIP T then we can look
into it :-)
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
Johansson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, April 05, 2006 3:37 PM
Subject: Re: [Asterisk-Users] SIP T
5 apr 2006 kl. 16.40 skrev Jon Weisman:
Anyone know how I can get SIP T working w/ Asterisk?
Start
Hi,
Anyone know of any active work to support SIP-T on Asterisk? Nortel is gaining ground in the carrier networks with CS2K where DMS is present, and are rumoured to have brought forward SIP and SIP-T support to mid year/3Q (GA dates).
Cheers
Rob
Rob Gillan
Director
DZhoN Pty. Ltd.
[EMAIL
[EMAIL PROTECTED] wrote:
But I would like asterisk to accept the IMTs.. and the only way to do
that is to find out what DS0 to take and place the calls on based on the
SS7 messages. I guess one of the things I'm not clear on is going from
SS7 to SIP-T, I'm not sure where the state machine
I, personally, think a channel driver handling both sigtran and
mgcp/megaco would be an ideal setup for bridging the gap between ip and
pstn.. especially with the current hardware devices on the market..
but all of that is just opinion..
Kevin P. Fleming wrote:
[EMAIL PROTECTED] wrote:
But I
Matt Hess wrote:
I, personally, think a channel driver handling both sigtran and
mgcp/megaco would be an ideal setup for bridging the gap between ip
and pstn.. especially with the current hardware devices on the market..
but all of that is just opinion..
Kevin P. Fleming wrote:
[EMAIL
Hey All,
I'm just daydreaming here.. but what's the status of SIP-T in Asterisk?
I haven't been able to find a whole lot of info on SIP-T but seems like
just an extension of SIP. Right?
Now if I had a PSTN Gateway (that is a SS7 gateway) that supported
SIP-T, could I signal * with SIP-T from
[EMAIL PROTECTED] wrote:
Just swiming around in it here.. Any thoughts? It seems to me that you
MUST use something like MGCP or H.248 to connect the call to the PSTN
(media gateway) since the specific DS0 to be utilized will be included
in the ISUP messages..
No, you can just do what you are
] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, January 24, 2005 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SIP-T Support (I got my head in an SS7
cloud)
Hey All,
I'm just daydreaming here.. but what's the status of SIP-T in
Asterisk?
I haven't
-
[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming
Sent: Monday, January 24, 2005 4:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP-T Support (I got my head in an SS7
cloud)
[EMAIL PROTECTED] wrote:
Just swiming around in it here.. Any
] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, January 24, 2005 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SIP-T Support (I got my head in an SS7
cloud)
Hey All,
I'm just daydreaming here.. but what's
But I would like asterisk to accept the IMTs.. and the only way to do
that is to find out what DS0 to take and place the calls on based on the
SS7 messages. I guess one of the things I'm not clear on is going from
SS7 to SIP-T, I'm not sure where the state machine exists..
-Brett
Kevin P.
Hi,
If I have two sip t.38 enabled gateways connected to asterisk, will I be
able to send a fax from one to the other with the mediastream passing
through asterisk ?
Thanks,
Dave
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