[Asterisk-Users] sip to sip

[Brian C. Fertig]

Well yes and no.  If they have static IP's then you only need to setup a
context as such: 

 

You would assign the following information on your Branch B server with
BranchA's information.

[branchA]

type=friend

defaultip=xxx.xxx.xxx.xxx

context=default

insecure=yes

host=xxx.xxx.xxx.xxx

disallow=all

allow=g729

allow=alaw

allow=ulaw

 

 

You would do the same here but for the Branch A server with Branch B's
config.

[branchB]

type=friend

defaultip=xxx.xxx.xxx.xxx

context=default

insecure=yes

host=xxx.xxx.xxx.xxx

disallow=all

allow=g729

allow=alaw

allow=ulaw

 

 

In your extensions.conf your dialplan would look something like this: 

 

exten => _30.,1,Dial(${EXTEN}@branchb,23,r)  ; use this for calling
people on branch B  

 

 

There is no need to register the boxes with each other if they are
static, which is the easiest way to set this up.

 

Any other questions lemme know.. 

 

.o-------------------------------------------------------o.

Brian Fertig

 

 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Quintin
Sent: Monday, 23 May, 2005 09:23
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] sip to sip 

 

Hi B

 

Do you mean I must do this in my sip.conf file on eatch server

 

Branch A 

register => 3001:1234 at 192.168.0.200 /3001

 

Branch B

register => 5001:1234 at 192.168.0.227 /5001

 

thx

Q

 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Brian C.
Fertig
Sent: 23 May 2005 03:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] sip to sip 

 

If your looking to link 2 asterisk boxes might I suggest IAX.  Much more
efficient in the way bandwidth

is utilized between the locations.  Also if you want to use your sip
solution, have you setup the other

end point in your SIP.CONF?   I have never got IP dialing to work in
asterisk but it works fine when

assigned in the conf file.

 

 

 

.o-------------------------------------------------------o.

Brian Fertig

NOC/Network Engineer

Systems Engineer

 

 

 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Quintin
Sent: Monday, 23 May, 2005 08:08
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] sip to sip 

 

Hi 

 

I'm trying to put up an sip pbx system for my company but i'm getting
some problems when I'm trying to call from server ( branch A ) to server
( branch B )...

 

This is my extentions.conf :

 

exten => 3003,1,Dial,SIP/3003 at 192.168.0.200

 

________________________________________________________

 

 

And this is what I get when I try to dial that user in branch B

 

_________________________________________________________

 

    -- Executing Dial("SIP/5001-66b1", "SIP/3003 at 192.168.0.200") in new
stack

    -- Called 3003 at 192.168.0.200

    -- Got SIP response 404 "Not Found" back from 192.168.0.200

    -- SIP/192.168.0.200-e638 is circuit-busy

  == Everyone is busy/congested at this time (1:0/1/0)

  == Auto fallthrough, channel 'SIP/5001-66b1' status is 'CONGESTION'

 

Both servers are exactly the same..... 

 

What can the problem be, that branch B server doesn't route the call
through

 

Thx

Quintin

________________________________

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