Two proxies scenario

Asterisk Asterisk Support
1

Hello again!

I have a very simple question.

I have:

  1. 2 SIP Proxies Proxy1 and Proxy1.
  2. 2 User Agents UA1, and UA2.
  3. UA1 registers to Proxy1 and UA2 registers to Proxy2.

Q:How to make a call from UA1 to the UA2 using two proxies?? I would like to make sure that the calls would always be set up through Proxy1 and then to Proxy2 to finaly arrive to the UA2.

How to configure files in ProxyA and ProxyB, which files must be changed??

Thanks,

Paroxyzm.

2

Hi
If ur talking proxy 1 & 2 is Asterisk , then it is matter of inetr asterisk exchange , So you need IAX channel for this.

Amit

3

[quote=“amit”]Hi
If ur talking proxy 1 & 2 is Asterisk , then it is matter of inetr asterisk exchange , So you need IAX channel for this.

Amit[/quote]

Yes both Proxy1 and Proxy2 are asterisk.

Sorry but im a complete newbie in asterisk stuff… its my first touch of it…

I did’nt understand a thing, so if you could put it more trivialy, I would be greatful.
Thanks,

Paroxyzm

4

Hi
that i was told you use IAX that is slotion for this. for that you need IAX provide. As u told me both proxy are Asterisk.
use IAX protocol with IAX proivider as we use in sip.

Amit

5

[quote=“amit”]Hi
that i was told you use IAX that is slotion for this. for that you need IAX provide. As u told me both proxy are Asterisk.
use IAX protocol with IAX proivider as we use in sip.

Amit[/quote]

Hello!

Could you tell me which files do I have to change in asterisk1 and asterisk2 to do that??
Also, what to write in sip.conf and extensions.conf ??
(iax.conf too??)

My files contain:

Asterisk1
@@@@@@@@@@@@@@@
sip.conf
#######
[general]
context=default
srvlookup=yes
port=5060
binaddr = 0.0.0.0

[caller]
type=friend
;secret=welcome
qualify=no
nat=no
host=dynamic
canreinvite=no
context=internal
username=caller

#######extensions.conf

[internal]
exten => callee,1,Dial(SIP/callee) //how to change this line? To “route” to Asterisk2 if there is no callee in localdomain?

Asterisk2
@@@@@@@@@@@@@@@
sip.conf
#######
[callee]
type=friend
;secret=welcome
qualify=no
nat=no
host=dynamic
canreinvite=no
context=internal
username=callee

extensions.conf
#######

exten => caller,1,Dial(SIP/caller) //how to change this line? To “route” to Asterisk1 if there is no caller in localdomain?

My original files (caller and callee in localdomain )were:

Asterisk_only_1
@@@@@@@@@@@@@@@
[general]
context=default
srvlookup=yes
port=5060
binaddr = 0.0.0.0

[caller]
type=friend
;secret=welcome
qualify=no
nat=no
host=dynamic
canreinvite=no
context=internal
username=caller

[callee]
type=friend
;secret=welcome
qualify=no
nat=no
host=dynamic
canreinvite=no
context=internal
username=callee

extensions.conf
[internal]
exten => callee,1,Dial(SIP/callee)
exten => caller,1,Dial(SIP/caller)

I hope I didnt make a mistake… I know this is not very difficult, but I’m complete newbie in asterisk.

As you can see this is the simple case, I would be very appreciated for any help.
Thanks,

Paroxyzm

6

Hi
i suggest go for voip-info.org & make search for this how to call 1 Asterisk to other u get all the deatils & follow that.
also go on asteriskguru.com.

Amit

7

[quote=“amit”]Hi
i suggest go for voip-info.org & make search for this how to call 1 Asterisk to other u get all the deatils & follow that.
also go on asteriskguru.com.

Amit[/quote]

Can someone post some GOOD trival examples???

I dont want to get into asterisk very much, but this extreamly simple case, that requires reading lots of things and keeps me frustrating.

Thanks,