[Asterisk-Users] Connecting 2 * Together-Pulling hair out
Chris
listmail at odisok.net
Thu May 5 14:45:44 MST 2005
I haven't gotten to keys yet.
The documentation out there doesn't seem to be very good.
Chris
----- Original Message -----
From: "Tim Pushor" <timp at crossthread.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
Sent: Thursday, May 05, 2005 4:06 PM
Subject: Re: [Asterisk-Users] Connecting 2 * Together-Pulling hair out
> Personally, if I owned both boxes and had full control of the dialplan
> on both, I'd stay away from passwords. (but be careful what I say, I'm a
> hack)
>
> I have a bunch of boxes connected together via IAX and authenticating
> via RSA. The entries in iax.conf are simple, and dialing across the
> connection is simple (no passwords in the dialplan) (thanks again Rich
> for taking the time).
>
> Tim
>
> Here is a sample of iax.conf entries on machine a:
>
> [machineb]
> type=user
> host=machineb.internal.net
> auth=rsa
> inkeys=machineb
> username=machineb
> context=inbound
>
> [machineb]
> type=peer
> host=machineb.internal.net
> auth=rsa
> outkey=machinea
> username=machinea
>
> And an example dialplan entry to dial an extention on machineb (in the
> inbound context):
>
> exten => 333,1,Dial(IAX2/machineb/333)
>
> And on machinea, the opposite of machineb:
>
> [machinea]
> type=user
> host=machinea.internal.net
> auth=rsa
> inkeys=machinea
> username=machinea
> context=inbound
>
> [machinea]
> type=peer
> host=machinea.internal.net
> auth=rsa
> outkey=machineb
> username=machineb
>
> To generate the keys:
>
> on machinea:
>
> astgenkey -n machinea
> mv machinea.* /var/lib/asterisk/keys
>
> copy machinea.pub to machineb's /var/lib/asterisk/keys
>
> on machineb:
>
> astgenkey -n machineb
> mv machineb.* /var/lib/asterisk/keys
>
> copy machineb.pub to machinea's /var/lib/asterisk/keys
>
>
> Chris wrote:
>
> > 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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> >>>Asterisk-Users at lists.digium.com
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> >>>
> >>>
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> >>------------------------------------------------------------------------
> >>
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