Hi to all,
I'm using asterisk 1.4.9 with chan_h323.
When someone in the H323-VoIP cloud dial 1234 this number is assigned to my
asterisk-machine, so the VoiceGW forward the flow to my machine, asterisk
though the dialplan can delivery the call to a particular SIP phone...this
is ok...
I can also dial from my sip phone every phone in the H323-VoIP cloud like
siemens....BUT...when I call to a cisco phone (model 7912) this start
ringing
asterisk*CLI>
> -- Executing [EMAIL PROTECTED]:1] Dial("SIP/user-08219f40", "
> H323/[EMAIL PROTECTED]|60)|Ttm") in new stack
> -- Requested transfer capability: 0x00 - SPEECH
> -- Called [EMAIL PROTECTED]
> -- Started music on hold, class 'default', on SIP/user-08219f40
> -- H323/XXX.XXX.XXX.XXX-10 is ringing
> -- H323/XXX.XXX.XXX.XXX-10 is ringing
>
> I answer and
== Everyone is busy/congested at this time (1:0/0/1)
> -- Stopped music on hold on SIP/user-08219f40
> == Auto fallthrough, channel 'SIP/user-08219f40' status is 'CHANUNAVAIL'
> asterisk*CLI>
>
> but when I call to cisco 7940 all thinghs function very well...problems
only with 7912...
any ideas???
bye
--
Alessandro R.
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