2 traditional PBXes with conflicting extensions, 1 * IP PBX

Asterisk Asterisk Support
1

Hi all,

Firstly, i’m experimenting for a few weeks with * and IP Hardphones. I’m getting the hang of it, but there’s one complicated thing I can’t figure out.

In this situation we’ve got 2 traditional PBXes (Sopho IS3000 & Hipath 3700). The Sopho is expanded with a SIP@NET card which operates like a SIP trunk with licenses for 60 extensions.

The Hipath 3700 has not yet been connect to the * server, but in the near future it will get connected also to the * server.

The extensions used by the Sopho range from 400 till 799. The ranges from the Hipath range from 700 till 950.

I’ve accomplished it to register the IP phones with * and at the same time at the Sopho SIP trunk. When someone dials for example 793 on the Sopho side, it forwards the call to * and it rings on the IP hardphone connected to *. Also calling from extension 793 (IP hardphone) to another (traditional) extension on the Sopho side works.

My problem is that i’ve got conflicting ranges (700 till 799 range) that exist on both traditional PBXes.

How do I configure this in extensions.conf or sip.conf to let the both traditional PBXes communicate with eachother via the * server?

I hope someone can help me! Thanks in advance!

Sebastiaan

2

You could use a prefix, example: * extension 1715 send the call to extension 715 of the Sopho, * extension 2715 send the call to extension 715 of the Hipath.

Bye.

Marco Bruni

3

Thnx for the advice!

I was already at the point that I should configure 2 extensions (in this situation 5715 (715) for Hipath extensions and 9715 (715) for Sopho extensions.

How do I specify this in extensions.conf? Do I need to configure one context (for example [mixed]) with both the extensions for Sopho and Hipath? And next to this different contexts for the extensions in both PBXes (e.g. [Sopho] and [Hipath]).

Can you describe this how to put the config in extensions.conf?

Sorry for maybe looking lazy but I tried al different things, but can’t get the hang of the different contexts.

Thanks in advance!
Sebastiaan

4

One context should be ok, let’s say the other two pbxs are connected to * via sip protocol and in sip.conf you defined then as sopho and hipath:

[mixed]

exten => _97XX,1,Dial(SIP/sopho/${EXTEN:1})
exten => _97XX,2,Hangup

exten => _57XX,1,Dial(SIP/hipath/${EXTEN:1})
exten => _57XX,2,Hangup

Bye.

Marco Bruni

5

Thanks for the info. I’ve put this in my extensions.conf:

[i][mixed]

exten => 5722,1,Dial(SIP/5722)
exten => 5722,2,Hangup()

exten => 73227,1,Dial(SIP/73227)
exten => 73227,2,Hangup()

exten => 7425,1,Dial(SIP/rick)
exten => 7425,2,Hangup()

exten => 36435,1,Dial(SIP/36435)
exten => 36435,2,Hangup()

exten => 227,1,Dial(SIP/227)
exten => 227,2,Hangup()

exten => _94XX,1,Dial(SIP/${EXTEN:1}@10.111.0.101)
exten => _94XX,2,Hangup()

exten => _95XX,1,Dial(SIP/${EXTEN:1}@10.111.0.101)
exten => _95XX,2,Hangup()

exten => _96XX,1,Dial(SIP/${EXTEN:1}@10.111.0.101)
exten => _96XX,2,Hangup()

exten => _97XX,1,Dial(SIP/${EXTEN:1}@10.111.0.101)
exten => _97XX,2,Hangup()[/i]

and in my sip.conf:

[i][5722]
type=friend
callerid=5722
host=dynamic
nat=yes
context=mixed

[73227]
type=friend
username=73227
nat=yes
caninvite=no
canreinvite=no
host=dynamic
context=mixed

[rick]
type=friend
username=rick
host=dynamic
context=mixed

[36435]
type=friend
username=36435
nat=yes
qualify=yes
caninvite=no
canreinvite=no
host=dynamic
context=mixed

[227]
type=friend
username=227
nat=yes
qualify=yes
caninvite=no
canreinvite=no
host=dynamic
context=mixed[/i]

I’ve got different users (external, internal). That’s why some of them are, for example, natted and others not.

My problem is that I can’t call from extension 5722 (number on IP hardphone) to extension 9700 (@ 10.111.0.101, number at Sopho PBX). It gives me an error that the user cannot be found.

What did I do wrong??

6

The error is on the * or sopho side ? Try post the * cli messages, this should help us understand what’s going on.

Bye.

Marco Bruni

7

Turn on sip debug and watch the output

8

I’ve got further now with the Sopho. It’s working with Asterisk, because we configured the IP card on the Sopho side like a trunk. We’ve got 30 channels into the Sopho so it’s working perfectly.

On the other side of the * box, we’ve got the Siemens Hipath 3700/3750. Its hooked up via an TMS2M card (same sort of card where the PSTN subscriber line is connected on), and on the * side it’s connected via an Digium TE120P. I’ve got the Digium card already working under Debian 4.1, through the Zaptel interface.

My next question is: How do I configure Zaptel and the Siemens Hipath to succesfull connect with eachother? I don’t get any line-sync… :frowning:

Thanks in advance!

9

I’ve got the physical link correct (needed another crosscable :smiley:). But now i’m confused about the different options on the Siemens Hipath side. There are 50+ options for which kind of provisioning is used on the Siemens side.

I understand that they have to match on both * and Siemens, but what do you guys suggest: Link based on QSIG, or link based on EuroISDN?

What I’m trying to configure, is a 30 channel trunk into and out of the Siemens to the * box, so that I can call analog phones on the Siemens side from * and can call from VOIP phones connected to * back to analog phones on the Siemens.

Thanks in advance!