Trunking 2 asterisk server with sip trunk

Asterisk Asterisk General
1

Hi,
I’m trying to establish the dial plan that trunking 2 asterisk server with sip trunk. Both server installed with
Digium AEX800.

                      AstA                                       AstB
                     ---------------                                ---------------
ext 100 -------- |             |                                |               |---------- ext 200
ext 101 -------- |             |         sip trunk          |                |---------- ext 201
ext 102  --------|             |------------------------------- |               |---------- ext 202
ext 103 ---------|             |                                |               |---------- ext 203
                     --------------                                 ---------------
                   ip=192.168.10.1                        ip=192.168.20.1

Ast A

/etc/asterisk/chan_dahdi.conf
[trunksgroups]

[channels]
usercallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0

;FXS Modules
group=1
signalling=fxo_ks
context=internal
chanel=1-4

;FX0 Modules
group=2
signalling=fxs_ks
context=incoming
channel=5-8

/etc/asterisk/sip.conf
[general]
register=>AstA:welcome@192.168.10.1/AstB
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
svrlookup=yes

[AstB]
type=friend
secret=welcome
context=AstB_incoming
host=dynamic
disallow=all
allow=ulaw

/ext/asterisk/extensions.conf
[global]
autofallthrough=yes

[internal]
exten=>100,1,Dial(DAHDI/1,20,rt)
exten=>101,1,Dial(DAHDI/2,20,rt)
exten=>102,1,Dial(DAHDI/3,20,rt)
exten=>103,1,Dial(DAHDI/4,20,rt)
exten=>_2XX,1,Dial(SIP/AstB/${EXTEN})

[AstB_incoming]
include=>internal

AstB

etc/asterisk/chan_dahdi.conf
[trunksgroups]

[channels]
usercallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0

;FXS Modules
group=1
signalling=fxo_ks
context=internal
chanel=1-4

;FX0 Modules
group=2
signalling=fxs_ks
context=incoming
channel=5-8

/etc/asterisk/sip.conf
[general]
register=>AstB:welcome@192.168.20.1/AstA
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
svrlookup=yes

[AstA]
type=friend
secret=welcome
context=AstA_incoming
host=dynamic
disallow=all
allow=ulaw

ext/asterisk/extensions.conf
[global]
autofallthrough=yes

[internal]
exten=>200,1,Dial(DAHDI/1,20,rt)
exten=>201,1,Dial(DAHDI/2,20,rt)
exten=>202,1,Dial(DAHDI/3,20,rt)
exten=>203,1,Dial(DAHDI/4,20,rt)
exten=>_1XX,1,Dial(SIP/AstA/${EXTEN})

[AstA_incoming]
include=>internal

Problem
I’m not able to make to call to other side of the asterisk server
Eg. extension 100 at the AstA to make a call to 200 at AstB

Please give me an advice on my Dial plan.

2

There are a couple of unnecessary complications in the sip.conf’s. Use static addresses and therefore remove the registers. Also make friend be peer.

Also set allowguest to no in the general sections. It’s a security risk, and any failure in the registration or authentication may still half succeed, as a result.

Then run with core set verbose 3 and provide the resulting trace output.

Also, this should have been on Asterisk Support.