Re: [asterisk-users] Pass through registration / proxy

2018-04-12 Thread TSG 
The challenge is that calls to extensions on the Nortel have to ring the
nortel phone AND the sip phone connected to Asterisk.  

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba
Sent: Wednesday, April 11, 2018 4:26 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Subject: Re: [asterisk-users] Pass through registration / proxy

On Wed, Apr 11, 2018 at 12:04:18PM -0400, Telium Technical Support wrote:
> Maybe proxy is the wrong word I chose.  Asterisk is something like a 
> peer to the legacy PBX.  I thought about setting up individual SIP 
> accounts on the Asterisk box to connect to the legacy PBX, or maybe a 
> SIP trunk to the legacy PBX (assuming it can route calls through the 
> SIP trunk to a peer to reach a phone).  The legacy PBX is a Nortel in case
that matters.
> 
> I'm supposed to figure this out and present options but having trouble 
> figuring out if Asterisk would be a peer, or pretend to be many sip 
> agents registering on the legacy Sip pbx, etc.  I think I'm stuck at 
> the conceptual level.  (Still a beginner in training - but having fun 
> learning Asterisk)

One of my first integrations was similar but with a Siemens. Easiest might
be a SIP trunk (peer) between Asterisk/Nortel and have different prefixes
for the Norted (1xx) and Asterisk (2xx) and route these to the other. The
SIP endpoints simply register to Asterisk.


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Re: [asterisk-users] Replacing PBX during a call in progress

2017-01-12 Thread TSG 
Can re-invites be sent AFTER the first Asterisk server has been shut down?  (If 
the first Asterisk server is still up then it’s a gracefull transition, but I’m 
assuming the first Asterisk server is simply unplugged).  And can they be sent 
from a NEW asterisk server?

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender
Sent: Thursday, January 12, 2017 12:06 PM
To: and...@telesip.net; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Replacing PBX during a call in progress

 

As Andres mentioned you can use VMWare. Another option would be to send a 
re-invite to both devices and send them to another server.

 

 

On Thu, Jan 12, 2017 at 12:03 PM, Andres  wrote:

On 1/12/17 11:09 AM, Telium Technical Support wrote:

This was asked many years ago but I thought I would check to see if things have 
changed.  Is it possible to take over a call in progress – using a replacement 
Asterisk server?  

One plausible scenario I can think of is if you are running VMware VMs.  Using 
the vMotion feature would accomplish subsecond VM live moves.



 

In other words, if 2 user agents are connected through an Asterisk PBX, and I 
tracked the call ID, IP of each UA (and anything else needed), could I remove 
the PBX and put a new one in its place (at the same IP address) and resume the 
call?  Somehow keeping the call up on the UA’s and telling Asterisk to just 
resume a call given specified parameters (so the UA’s wouldn’t notice the 
change)?

 

 

 

 






-- 
Technical Support
http://www.telesip.net


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Re: [asterisk-users] Replacing PBX during a call in progress

2017-01-12 Thread TSG 
That's the same VM guest moved to a different VM host (not really what I was
looking forward).  In this case it's an entirely new host with Asterisk
having no state/session information, but my app would repopulate the session
info and try to re-establish the call.

 

Given SIP over TCP I suspect the answer is still now (since opening the
connection on a new host would result in a new syn handshake, different
source port used by Asterisk etc.)

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
Sent: Thursday, January 12, 2017 12:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Replacing PBX during a call in progress

 

On 1/12/17 11:09 AM, Telium Technical Support wrote:

This was asked many years ago but I thought I would check to see if things
have changed.  Is it possible to take over a call in progress - using a
replacement Asterisk server?  

One plausible scenario I can think of is if you are running VMware VMs.
Using the vMotion feature would accomplish subsecond VM live moves.



 

In other words, if 2 user agents are connected through an Asterisk PBX, and
I tracked the call ID, IP of each UA (and anything else needed), could I
remove the PBX and put a new one in its place (at the same IP address) and
resume the call?  Somehow keeping the call up on the UA's and telling
Asterisk to just resume a call given specified parameters (so the UA's
wouldn't notice the change)?

 

 

 










-- 
Technical Support
http://www.telesip.net
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