Good to hear about progress! I currently don't have access to a Cisco router
with FXO interface, so there might be errors here.
Now, if you only use _one_ FXO port, to my knowledge, you can only forward it
to _one_ extension. So, let's say that you configure
FXO-port 1/1/0 with plar 101, then all incoming PSTN-calls on that FXO
interface will end up with a SIP INVITE to sip:1...@sipserver. If
you wanna connect to another SipX "extension", you need multiple FXO interfaces
with different plar commands. However, forwarding
the call to an auto attendant on SipX should absolutely work...
Now, for the outgoing part. You need a dial-peer that matches you outbound call
leg from SipX to the FXO interface. So, if you have
a SIP trunk configured in SipX that points to the Cisco router for a specific
dial plan rule, the digits you send on that trunk must have
a matching dial-peer on the router, connected to the FXO interface.
Your dial-peer 100 voip matches the plar 101 and sends it to SipX. I would
configure the destination-pattern to explicitly match
the plar, but that's just me.
Now, your dial-peer 10 is a pots dial-peer matching 8 digits starting with a
'9'. My guess is that this is what you send from
SipX on the trunk? Is this really the digits you send to SipX? pots dial-peers
usually (i'm not really sure about this) perform
'pre-dot-strip', so the '9' would be stripped before sending out on the FXO
interface anyhow, even without the 'forward digits 7' command.
Dial-peer 101 voip, this would be the dial-peer matching the incoming SIP
INVITE from SipX that is supposed to be sent out
on the FXO interface. Now, I would use the "incoming-called number" command
here to match instead of the "destination-pattern".
So, based on the previous dial-peer, "incoming-called number 9......."
A good debug command to verify the SIP signaling in the router is:
router# debug ccsip messages
##### Rough out-of-the-head EXAMPLE #####
PSTN: FXO 1/1/0, subscriber 555-12345
Router IP: 10.10.10.2
SipX IP: 10.10.10.1
Prefix from SipX for PSTN access: '9'
Attendant extension on SipX, target for incoming PSTN calls: 100
!
voice-port 1/1/0
cptone SE
description *** FXO port 1, PSTN 555-12345 ***
port 1/1/0
plar 100
!
!
dial-peer voice 10 voip
description *** SIP trunk from router to SipX ***
destination-pattern 100
voice-class codec 1
session protocol sipv2
sesstion target ip:10.10.10.1
dtmf-relay rtp-nte
fax protocol pass-through g711alaw
no vad
!
!
dial-peer voice 20 voip
description *** SIP trunk from SipX to router ***
incoming-called number 9.T
voice-class codec 1
sesstion protocol sipv2
dtmf-relay rtp-nte
fax protocol pass-through g711alaw
no vad
!
!
dial-peer voice 21 pots
description *** Outbound to PSTN using FXO ***
incoming-called number 9.T
port 1/1/0
!
!
######
Best regards,
/Staffan
--
Staffan Kerker
mail/sip/xmpp: [email protected]
"Don't get involved in politics man, just play the gig..." /Sgt Floyd, Electric
Mayhem Band
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