One of the things I'm trying to do it to connect my 8x8 DTA 310 terminal
adapter onto my asterisk. I have the 8x8 box connected to the Internet, and
the phone line connected to an fxo port on a Cisco router:
voice-port 0/2/0
connection plar opx 5000
caller-id enable
dial-peer voice 200 voip
destination-pattern 5...
session protocol sipv2
session target sip-server
codec g711ulaw
!
sip-ua
sip-server ipv4:172.16.200.212 <------ Asterisk server
When I make a call from the PSTN to the 8x8 box, it does send ring back to the
asterisk server and the Digium phone does ring. However, as soon as the phone
rings the call disconnects yet the actual phone, extension 5000, rings two
times before it hangs up, also.
The following output is what I see on the Asterisk console:
asterisk*CLI>
== Using SIP RTP CoS mark 5
[Oct 18 16:27:46] NOTICE[1513]: chan_sip.c:23352 handle_request_invite: Call
from '' (172.16.200.1:65451) to extension '5000' rejected because extension not
found in context 'default'.
== Using SIP RTP CoS mark 5
-- Executing [5000@pstn-incoming:1] Dial("SIP/172.16.200.1-00000006",
"SIP/5000,20|p") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/5000
-- SIP/5000-00000007 is ringing
== Spawn extension (pstn-incoming, 5000, 1) exited non-zero on
'SIP/172.16.200.1-00000006'
The 172.16.200.1 is my router.
sip.conf excerpt:
[5000]
type=friend
context=phones
host=dynamic
disallow=all
allow=ulaw
secret=cisco123
mailbox=5000@phones
[172.16.200.1]
context=pstn-incoming
type=friend
host=172.16.200.1
dtmfmode=rfc2833
disallow=all
allow=ulaw
[phones]
exten => 5000,1,Dial(SIP/${EXTEN},20|p)
exten => 5000,n,Hangup
[pstn-incoming]
include=phones
Any help would be greatly appreciated,
Thanks,--
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