Re: [asterisk-users] How to set up Asterisk to deliver a trunk sip connection?

2010-08-11 Thread David Backeberg
On Wed, Aug 11, 2010 at 11:24 AM, Kent Varmedal  wrote:
> We need to upgrade this PBX for it to work with SIP, it is at the moment
> using ISDN. And those who delivered it and do the
> support/reconfiguration is paid by the hour. We don't have any control
> over it our self, so when it is changed it will stay that way.

If you have a spare ISDN card, you may prefer to integrate it over
T1/E1/PRI rather than over SIP.

The bigger barrier is you not having control of the system.

It's made a lot tougher by not having a way to experiment except to be
dead-in-the-water.

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] How to set up Asterisk to deliver a trunk sip connection?

2010-08-11 Thread Kent Varmedal
On Wed, 2010-08-11 at 10:29 -0400, David Backeberg wrote:
> On Wed, Aug 11, 2010 at 10:12 AM, Kent Varmedal  wrote:
> > I'm trying to set up an "old" PBX (that supports SIP) to go through our
> > new Asterisk server, so that our old phones can be used still for some
> > time.
> >
> > How can I set up Asterisk to deliver a trunk sip connection that our old
> > PBX can connect to? Is it just to sett up a normal sip device in
> > sip.conf? Or is there some other / extra magic for this to work?
> 
> Pretty much. The details vary on codec / DTMF, etc. based on what
> you're talking to, but that's the general idea.
> 
> > I can't test this with the old system before we go live with Asterisk
> > (and then it must work).
> 
> Really? Why not? If it speaks SIP, it should be able to do multiple
> SIP trunks / channels, and you should be able to set up a simultaneous
> SIP trunk alongside your production line(s).
> 

We need to upgrade this PBX for it to work with SIP, it is at the moment
using ISDN. And those who delivered it and do the
support/reconfiguration is paid by the hour. We don't have any control
over it our self, so when it is changed it will stay that way.

> If you tell us the PBX, somebody here has probably worked with it.
> 

It is a Ascotel intelliGate 2025/2045.

> If the old PBX speaks true SIP, you could ditch the old PBX, and have
> the SIP phones register directly with asterisk.
> 

We don't have any SIP phones at the moment, we use DECT phones from
Aastra (not the IP version). New phones will probably be SIP. 


Best regards,
Kent Varmedal




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] How to set up Asterisk to deliver a trunk sip connection?

2010-08-11 Thread David Backeberg
On Wed, Aug 11, 2010 at 10:12 AM, Kent Varmedal  wrote:
> I'm trying to set up an "old" PBX (that supports SIP) to go through our
> new Asterisk server, so that our old phones can be used still for some
> time.
>
> How can I set up Asterisk to deliver a trunk sip connection that our old
> PBX can connect to? Is it just to sett up a normal sip device in
> sip.conf? Or is there some other / extra magic for this to work?

Pretty much. The details vary on codec / DTMF, etc. based on what
you're talking to, but that's the general idea.

> I can't test this with the old system before we go live with Asterisk
> (and then it must work).

Really? Why not? If it speaks SIP, it should be able to do multiple
SIP trunks / channels, and you should be able to set up a simultaneous
SIP trunk alongside your production line(s).

If you tell us the PBX, somebody here has probably worked with it.

If the old PBX speaks true SIP, you could ditch the old PBX, and have
the SIP phones register directly with asterisk.

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users