Authentication problems should be displayed. I don't use passwords (the -n flag to astgenkey). The name of the key needs to be whatever you put in inkeys.

Did you try using the configuration from my original email verbatim? How about showing us the relavent portions of the iax.conf and dialplan, as well as details as to where you put your keys, both private and public.


Chris wrote:

   I have.    The receiving box doesn't show anything, but tcpdump shows 
bidirectional traffic.
I can only think that it is an authentication problem.    When you generate the 
key do you need the password?   Does the name of the key have to match the user 
name?

Chris

----- Original Message ----- From: "Tim Pushor" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]>
Sent: Friday, May 06, 2005 12:33 PM
Subject: Re: [Asterisk-Users] Connecting 2 * Together-Pulling hair out





What does your log/debug tell you?

Set debug and log levels up, and watch the fun.

Chris wrote:



  I tried it that way, but it just rings and eventually says all circuits are 
busy.


Chris

----- Original Message ----- From: "Tim Pushor" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]>
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" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[email protected]>
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[email protected]>
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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