Hi everybody
Could someone give a tip on how can I configure asterisk to use 2 ATA's
186 to communicate each other using SIP with asterisk. I know this most
be a very simple task, however this is the very first aproach I have to
asterisk. I set the following config but I don't get dial-tone when I
off-hook the phone from any of the two ATAs. Can some one tell what I'm
missing in the configuration??
sip.conf file
[general]
port = 5060 ; Port to bind to
bindaddr = 192.168.0.254 ; Address to bind to
context = default ; Default for incoming calls
tos=lowdelay
;tos=184
maxexpirey=3600 ; Max length of incoming registration we allow
defaultexpirey=120 ; Default length of incoming/outoing
registration
;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY
;
;register => 9873@192.168.0.2 ; Register with a SIP provider
;register => 9874@192.168.0.2
;register => 2345@148.243.196.14/1234 ; Register 2345 at sip provider as
1234 here.
;allow=g729
;
;[cisco]
type=friend
username=9873
secret=pwd
;nat=yes ; This phone may be natted
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=200 ; Qualify peer is no more than 200ms away
defaultip=192.168.0.5
mailbox=9873
;[cisco2]
type=friend
username=9874
secret=pwd
nat=yes ; This phone may be natted
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=200 ; Qualify peer is no more than 200ms away
defaultip=192.168.0.10
mailbox=9874
extensions.conf
I added this at the end of the extension.conf file:
exten => 9873,1,Dial(SIP/cisco,30,tr)
exten => 9873,2,Playback(new/nbdy-avail-to-take-call)
exten => 9873,3,Voicemail(u9873)
exten => 9873,4,Hangup
exten => 9873,102,Voicemail(b9873)
exten => 9873,103,Hangup
exten => 9874,1,Dial(SIP/cisco2,30,tr)
exten => 9874,2,Playback(new/nbdy-avail-to-take-call)
exten => 9874,3,Voicemail(u9874)
exten => 9874,4,Hangup
exten => 9874,102,Voicemail(b9874)
exten => 9874,103,Hangup
And that's all I did. However I'm not sure If I have to configure
something else?? I also have a SIP proxy server(Not asterisk) and I
pretend to send out calls through this proxy server, but thisonce the 2
ATAs can call to each other behid asterisk.
Can someone give a hint on this??? Any tip would be appreciated.
I'm actually using Redhat 9. The ATAs are using the 2.16 firmware. The
ATA's are pointing to the asterisk bind address 192.168.0.254 in the
sip.conf
Thanks in advance!!
Kind Regards!!
This is what I have:
ATA 1, UID0=9873
192.168.0.5-------------
|
|----------Asterisk
BOX---------------------SIP-Proxy Server
ATA 2, UIDO=9874 | 192.168.0.254
192.168.0.2
192.168.0.10 -----------
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