trunk between 2 servers

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Asterisk Users 2 Comments

Hello,

We have 1 server installed with centos and asterisk and the sip in
configured we can do the external and internal call without issue

Now I have installed the same centos and asterisk 1.4 in second computer in
the same network.

My question if there is any possibility to create a trunk between the first
and the second PC in oorder do the inbound and outbound calls

Thanks and Regards

2 thoughts on - trunk between 2 servers

  • Sure.

    Depending on your needs, DUNDI may be appropriate.

    For my needs (1 man band), I just use IAX. SIP will also do the trick,
    it’s just a little bit more complex to set up but should be well within
    your skill set if you already have SIP configured for [in/out]bound on
    your first box.

    After you’ve configured iax.conf correctly, all you need is something
    like:

    exten = *,n,dial(iax2/user:pass@second/exten-in-second-dialplan)

    In the ‘first’ box’s dialplan.

  • Hi!

    Agree on the IAX. Had never used it, I was a SIP-Only one man band. Not
    got a VPS in a different location and configured IAX between them. Cool
    thing is it has encryption built-in, in case you don’t have both of the
    machine locally.

    As a really ugly fail-over setup I just made a new context in my
    extentions.conf for iax-in where I also have all my extentions (like all
    6! :-). In the sip context in the dialplan I added an entry to dial
    extention@other_iax node in case of failure.

    So now I get trunked calls in on one of both boxes (you can setup an SRV
    to configure the standby machine) and I’ll try the ext locally. If that
    one fails (not registered) it’ll send the call over the IAX trunk to the
    other machine. Fails there: voicemail.

    Basically there’s no end to what you can do this way, like have some
    trunk operators connected to the machine that’s closest by and deliver
    the calls there yourself over IAX. IAX can have monitoring and
    encryption, pretty nice stats too. If you managed to setup SIP, IAX is
    easy. I only used Google basically. Just don’t forget the two contexts
    to prevent loops. And in my ugly work-around you’ll have two places to
    check voicemail, if you would use that.

    /Robin