[Asterisk-Users] Connecting 2 * Together-Pulling hair out

Thu May 5 13:10:13 MST 2005

    I have something similar.  Both of my servers are behind a firewall and NAT.  You will need to allow UDP 4569 through the firewall for IAX2. If you have NAT you will need to redirect 4569 to the internal server.  

    I would suggest using AMP and then looking at IAX_ADDITIONAL.CONF to see how it's done. You can modify the IAX.CONf because I don't believe AMP rewrites that file.

    I think the user and passwords are required.   I would suggest using a strong password or someone may decide to make a few phone calls.   After this you will need the routing in Extensions.conf to allow calls to be made on this trunk.

    Asterisk will handle the SIP > IAX.    All my clients are SIP and they have no trouble going over a IAX trunk to other SIP devices on the other server.

This is what my IAX_ADDITIONAL.CONF looks like

SiteA - Dynamic IP
--------------
[boxb-peer]
username=boxa-user
type=peer
trunk=yes
secret=mypassword
host=thehost.dyndns.org

[boxb-user]
type=user
secret=mypassword2
host=thehost.dyndns.org
context=from-internal

---------------
Site b - Static IP
----------------

[boxa-peer]
username=boxb-user
type=peer
trunk=yes
secret=mypassword2
host=xxx.xxx.xxx.xxx

[boxa-user]
type=user
secret=mypassword
host=xxx.xxx.xxx.xxx
context=from-internal


Regards,

Chris


----- Original Message ----- 
From: "mr. barker" <cabalitomb at shaw.ca>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users at lists.digium.com>
Sent: Thursday, May 05, 2005 1:58 PM
Subject: RE: [Asterisk-Users] Connecting 2 * Together-Pulling hair out


> Yes trying to connect to boxes together.
> 
> One sits outside the internal firewall and is on the inside.
> 
> I am using AMP.  However I can just put whatever I need in the custom.conf
> sections.
> The users agents are SIP .. can SIP call go over a IAX trunk ? if so great.
> To create the trunk do I need to use a users name and password ? or ?
> 
> I need to have the *box that is behind the firewall to be able to place a
> call out through the *box that has a public ip.
> 
> Thank you
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Chris
> Sent: Thursday, May 05, 2005 8:20 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Connecting 2 * Together-Pulling hair out
> 
>     I am not sure what you are trying to do.    I have created an IAX2 trunk
> between the servers over an internet connection.
> Then all you have to do is put in call routing on the trunks to forward the
> call to the right place.  Are you using AMP or trying to do it manually.
> I found everything a little confusing as well, but it is simple now that I
> understand it.
> 
> 
> Chris
> 
> ----- Original Message ----- 
> From: "mr. barker" <cabalitomb at shaw.ca>
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Sent: Thursday, May 05, 2005 4:43 AM
> Subject: [Asterisk-Users] Connecting 2 * Together-Pulling hair out
> 
> 
> > 
> > 
> >  
> > 
> >   _____  
> > 
> > Subject: [Asterisk-Users] Connecting 2 * Together-Pulling hair out
> > 
> >  
> > 
> > I have read the docs on connecting 2* together but am unsure of a few
> things
> > 
> >  
> > 
> > Do I need a different account for each number that will be called from one
> > box to the other ? ie. Do I set up a user account on one and then have the
> > other box log into that account when it whats to make a call ?
> > 
> >  
> > 
> > I have 2 asterisk boxes and only one of them has the ability to access a
> > VoipAccount and PSTN connections.(*box 1). The other holds the SIP
> > extensions for the internal SIP users/exten(*box2)
> > 
> > I would like to be able to have the box with the Sip UA(*box2) on it to be
> > able to place a call using the box that has the VoipAccount and PSTN
> > connection.  I am able to make multiple UA calls on the VoipAccount and 3
> on
> > the PSTN lines (only have 3 lines coming in).  I can get it to work if I
> > create a user exten on *box1 and map a trunk(which is really only an
> exten)
> > using the user/password login to that exten from *box2.  However when I
> try
> > to place a second call when the VOIP line is in use it gives me error (
> > basically saying can't use the trunk because it is in use)  I would like
> to
> > be able to have this exten/trunk to be able to use multiple connections on
> > it.
> > 
> >  
> > 
> > There must be an easier way to do this I am just not sure how.  I looked
> at
> > creating IAX trunks but still come up with the Trunk is really an Exten
> > name/password .  
> > 
> >  
> > 
> > Any help would be appreciated. (my brain is boiling eggs)
> > 
> >  
> > 
> > Thank you.
> > 
> >  
> > 
> >  
> > 
> >  
> > 
> > 
> 
> 
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