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Hi everybody<br>
<br>
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??<br>
<br>
<b>sip.conf file</b><br>
<br>
[general]<br>
port = 5060 ; Port to bind to<br>
bindaddr = 192.168.0.254 ; Address to bind to<br>
context = default ; Default for incoming calls<br>
tos=lowdelay<br>
;tos=184<br>
maxexpirey=3600 ; Max length of incoming registration we allow<br>
defaultexpirey=120 ; Default length of incoming/outoing registration<br>
;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY<br>
;<br>
;register => <a class="moz-txt-link-abbreviated" href="mailto:9873@192.168.0.2 ">9873@192.168.0.2 </a> ; Register with a SIP provider<br>
;register => <a class="moz-txt-link-abbreviated" href="mailto:9874@192.168.0.2">9874@192.168.0.2</a><br>
;register => <a class="moz-txt-link-abbreviated" href="mailto:2345@148.243.196.14/1234">2345@148.243.196.14/1234</a> ; Register 2345 at sip provider
as 1234 here.<br>
;allow=g729<br>
;<br>
;[cisco]<br>
type=friend<br>
username=9873<br>
secret=pwd<br>
;nat=yes ; This phone may be natted<br>
host=dynamic<br>
canreinvite=no ; Cisco poops on reinvite sometimes<br>
qualify=200 ; Qualify peer is no more than 200ms away<br>
defaultip=192.168.0.5<br>
mailbox=9873<br>
<br>
;[cisco2]<br>
type=friend<br>
username=9874<br>
secret=pwd<br>
nat=yes ; This phone may be natted<br>
host=dynamic<br>
canreinvite=no ; Cisco poops on reinvite sometimes<br>
qualify=200 ; Qualify peer is no more than 200ms away<br>
defaultip=192.168.0.10<br>
mailbox=9874<br>
<br>
<br>
<b>extensions.conf</b><br>
I added this at the end of the extension.conf file:<br>
<br>
exten => 9873,1,Dial(SIP/cisco,30,tr)<br>
exten => 9873,2,Playback(new/nbdy-avail-to-take-call)<br>
exten => 9873,3,Voicemail(u9873)<br>
exten => 9873,4,Hangup<br>
exten => 9873,102,Voicemail(b9873)<br>
exten => 9873,103,Hangup<br>
<br>
<br>
exten => 9874,1,Dial(SIP/cisco2,30,tr)<br>
exten => 9874,2,Playback(new/nbdy-avail-to-take-call)<br>
exten => 9874,3,Voicemail(u9874)<br>
exten => 9874,4,Hangup<br>
exten => 9874,102,Voicemail(b9874)<br>
exten => 9874,103,Hangup<br>
<br>
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. <br>
<br>
Can someone give a hint on this??? Any tip would be appreciated.<br>
<br>
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<br>
<br>
Thanks in advance!!<br>
<br>
Kind Regards!!<br>
<br>
This is what I have:<br>
<br>
ATA 1, UID0=9873<br>
192.168.0.5-------------<br>
|<br>
|----------Asterisk BOX---------------------SIP-Proxy
Server<br>
ATA 2, UIDO=9874 | 192.168.0.254 192.168.0.2<br>
192.168.0.10 -----------<br>
<br>
<br>
<br>
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