I have sucessfully configured and connected the two asterisk servers.
For tht, considre the following scenario.
there are two pcs each have asterisk server installed. (asterisk - 1.2.24)
there are two sip fones also, first fone(fon no 5000) and second fon(fon no 5010).
for instance I am giving the sample ips to tht pcs like
(1) pc1 : IP - 192.168.100.200 Port - 5060
(2) pc2 : IP - 192.168.100.201 Port - 5060
fon_1 configuration,
---------------------------------------------
SIP Server : 192.168.100.200:5060
SIP User ID: 5000
Authenticate ID:5000
Name:5000
----------------------------------------------
fon_2 configuration,
---------------------------------------------
SIP Server : 192.168.100.201:5060
SIP User ID: 5010
Authenticate ID:5010
Name:5010
----------------------------------------------
Now u just have to edit 3 files. (in both the pcs)
-------------------------------------------------------------------
pc_1 conf files:
------------------------------------------------------------------
sip .conf------------------------------------------
[default]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
[5000]
type=friend
username=5000
authuser=5000
context=iax
host=dynamic
canreinvite=no
nat=no
disallow=all
allow=g729
sip.conf file over -------------------xxxx-----------------
iax.conf ------------------------------------------
;peer name
[ASTtest1]
type=friend
context=default
auth=plaintext
host=192.168.100.201
port=4569
permit=0.0.0.0/0.0.0.0
disallow=all
allow=g729
iax.conf file over -----------------xxxx-----------------------
extensions.conf----------------------------------
[iax]
exten => _5XXX,1,Dial(IAX2/ASTtest1/${EXTEN},55,o)
exten => _5XXX,n,hangup()
extensions.conf file over --------------xxxxx--------------
-------------------------------------------------------------------
pc_2 conf files:
------------------------------------------------------------------
sip .conf------------------------------------------
[default]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
[5010]
type=friend
username=5010
authuser=5010
context=iax
host=dynamic
canreinvite=no
nat=no
disallow=all
allow=g729
sip.conf file over -------------------xxxx-----------------
iax.conf ------------------------------------------
;peer name
[ASTtest2]
type=friend
context=default
auth=plaintext
host=192.168.100.200
port=4569
permit=0.0.0.0/0.0.0.0
disallow=all
allow=g729
iax.conf file over -----------------xxxx-----------------------
extensions.conf----------------------------------
[iax]
exten => _5XXX,1,Dial(IAX2/ASTtest2/${EXTEN},55,o)
exten => _5XXX,n,hangup()
extensions.conf file over --------------xxxxx--------------
all configuration stuff is over (cheers !!!
)
now if u want to check in sip server then u can fire the following command.
CLI> iax2 show peers
u can make calls also from 5000 to 5010. (If u find not working fons then just check for codek compatibility in fone)
its time to rock an rollllll, fun unlimited........................................................
