Thanks with the upgrade they work... Now i only have one problem.<br>
<br>
I create 3 hints.. (111 (SIP/111), 112 (SIP/112), and 102 (ZAP/35) )<br>
the SIP/111 is a GrandStream ATA<br>
the SIP/112 is a Polycom 301<br>
the ZAP/35 is a Analogic Phone.<br>
<br>
The SIP/112 hints works great. But the other 2 no.<br>
<br>
The ZAP/35 is say is always in USE and as you see en the<br>
next console output is not in use. any Idea????<br>
<br>
asterisk*CLI><br>
-= Registered Asterisk Dial Plan Hints =-<br>
111
:
SIP/111
State:Idle
Watchers 4<br>
102
:
ZAP/35
State:InUse
Watchers 5<br>
112
:
SIP/112
State:InUse
Watchers 2<br>
----------------<br>
- 3 hints registered<br>
asterisk*CLI> show cha<br>
channel channels channeltypes<br>
asterisk*CLI> show channels<br>
Channel
Location
State Application(Data)<br>
Zap/34-1
s@incel:1
Up Bridged Call(SIP/112-1f3d)<br>
SIP/112-1f3d
90443338182842@home: Up
Dial(ZAP/34/3338182842|120|Tt)<br>
2 active channels<br>
1 active call<br>
<br>
And also the SIP/111 is always in Idle any idea of why ???<br>
<br>
thanks<br>
<br><br><div><span class="gmail_quote">On 11/9/05, <b class="gmail_sendername">Peter Dean</b> <<a href="mailto:peter.john.dean@gmail.com">peter.john.dean@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Upgrade to asterisk-1.2.0-rc1 and ensure that your sip file contains<br>subscribecontext=sip-text.<br>_______________________________________________<br>--Bandwidth and Colocation sponsored by <a href="http://Easynews.com">
Easynews.com</a> --<br><br>Asterisk-Users mailing list<br><a href="mailto:Asterisk-Users@lists.digium.com">Asterisk-Users@lists.digium.com</a><br><a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users
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