Hi to all,
I am using asterisk-1.4.21.1 ,asterisk-gui-2.0.4 with cent OS 5.4 server. I am working in the call flow from asterisk GUI.I created the trunk like this in users.conf in first server
[Trunk]
host = 10.40.10.9
username = root
secret = ucs123
trunkname = Trunk ; GUI metadata
type=peer
context = DID_Trunk
hasexten = no
hasiax = no
hassip = yes
registeriax = no
registersip = yes
trunkstyle = voip
insecure = very
dtmfmode=rfc2833
disallow = all
allow = ulaw,alaw,g729,ilbc
fromuser=6004
fromdomain=10.40.10.8
qualify=yes
nat=auto
In second server users.conf is like this
[Trunk]
host = 10.40.10.8
username = root
secret = root12
trunkname = Trunk ; GUI metadata
type = peer
context = DID_Trunk
hasexten = no
hasiax = no
hassip = yes
registeriax = no
registersip = yes
trunkstyle = voip
insecure =
dtmfmode = rfc2833
fromuser = 6000
fromdomain = 10.40.10.9
qualify = yes
nat = auto
disallow = all
allow = ulaw,alaw,g729,ilbc
when i try to make a call from server1 to server2 it showing the message like this
Executing [6000@DLPN_DialPlan1:1] Dial("SIP/6004-b7512ce0", "SIP/6000") in new stack
[Apr 26 08:47:45] WARNING[1504]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/6004-b7512ce0' status is 'CHANUNAVAIL'
can any body help me what is the error in this
Thanks&Regards,
Voip.