Dial extension on another PBX server

Hi, I have two PBX (asterisk) servers. How can I easily create a dial plan that will call up a device on a different PBX server. Thanks!

Asterisk sees this as no different from calling a PSTN number via a network operator.

What does that mean? Can I not do what I’m trying to do easily?

Nearly every user of Asterisk has no difficulty configuring it to make calls through an ITSP or an analogue or ISDN phone line direct to a PTT, so I’m saying it is so easy, that, once you stop thinking it is something special, it is not something you need to ask about.

Create peers for each machine, Dial the peer and the extension.

Assuming server1’s handsets are in the 100 range, and Server 2’s handsets are in the 200 range it’d look vaguely something like this.

I wrote this all off the top of my head, I recommend reading the Wiki to learn more.

[server1-internal]
exten => _1XX,1,Dial(SIP/${EXTEN})
exten => _2XX,1,Dial(SIP/server2/${EXTEN})

[server2-internal]
exten => _1XX,1,Dial(SIP/server1/${EXTEN})
exten => _2XX,1,Dial(SIP/${EXTEN})

[server1-external]
exten => _1XX,1,Dial(SIP/${EXTEN})

[server2-external]
exten => _2XX,1,Dial(SIP/${EXTEN})


[server1]
type=friend
host=10.10.10.1
context=server1-external
allow=ulaw
qualify=yes

[server2]
type=friend
host=10.10.10.2
context=server2-external
allow=ulaw
qualify=yes
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Is there anyway you could point out where in the asterisk documentation it shows how do set this up? I’ve been pulling my hair out reading the document and trying to figure it out. Thank you so much.

Firstly, nowhere did you say you were using SIP, or even VoIP, however, for SP:

http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics-SECT-3.html#DialplanBasics_id292362

Also, at the start of sip.conf.sample, which is essential reading for anyone using SIP beyond very basic GUI use (this forum doesn’t cover GUI use). This is in the source tree and all the sample files should be included with any package intended for direct configuration, as they are, basically the documentation…

There are also examples, for both SIP and DAHDI, in extension.conf.sample, which is also essential reading.

I apologize, I am using SIP. I also did read the sample sip.conf file that was installed, but it didn’t really shed any light on how to do this considering it is over 1600 lines. Do you think you could point out which portion of sip.conf.sample sheds light on how to do what I’m trying to? Thank you again for your help

Could you specify where all of this goes? Does it go on server1 or server2? Does it go in both?

I’m afraid I don’t do paid consultancy at the moment. Whilst I’ll answer questions here, I’ll only do so to point people in the right direction, so that they can fill in small gaps in their knowledge.

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http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/OutsideConnectivity_id291235.html#OutsideConnectivity_id291281 Check “Configuring SIP trunks between Asterisk systems”

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I provided example configurations for both. Read @satish4asterisk 's link.