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

Tim Pushor Thu, 05 May 2005 16:53:16 -0700

You don't need to have multiple connections defined. (I think you can IAX trunk multiple conversations over ONE IAX connection (one data stream) but you can run multiple streams without problem (if you had concurrent multiple connections you may be able to gain some efficiency using IAX trunking).

I am batching it this weekend and probably will be hacking away at my setup, so if you want, give me a call at fwd #561293, ext 100 if you are pulling your hair out ;-)

Tim

mr. barker wrote:

Thank you to both Chris and Tim
I could not get my head around this .. after seeing the examples it now
makes sense what needs to be done.  I will give both a whirl tonight.
I do like the RSA key idea.
One question is this, will I need multiple accounts on the Static IP
machines so the Dynamic machine has the ability to make more then one
concurrent SIP call through the Static IP machine ?
If I could get the Static IP box to go through the my SMC router it would be
great.  I tried opening the ports. 5060udp/tcp, 10000-20000udp/tcp.
Tried even setting the machine in the DMZ zone.  I think the VOIP provider
just has problems translating through the NAT or something.
The linux box is running [EMAIL PROTECTED] no firewall setting that I know of.
To much of a Newbie at linux .. lol and I have been at it for almost 1 year
now and still have soooo much to learn.
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Sent: Thursday, May 05, 2005 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Connecting 2 * Together-Pulling hair out
   I haven't gotten to keys yet.
The documentation out there doesn't seem to be very good.
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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