01-08-2012 01:05 AM - edited 03-16-2019 08:52 AM
Hi All,
I have calls coming via SIP trunk in a 2611xm router that has 4 FXO lines for dialing out to campus exchange. The setup is working for both inbound and outbound calls. Users are dialing a standard pattern e.g. XXXX , 9XXX XXXX etc.
Now is there a way to restrict incoming SIP call i.e. from the calling number e.g. 1000 to go out via a specific FXO port, while rest of the the calls should go out via remaining 3 FXO ports.
Thanks
-Azher
Current Config:
>>>> Four Analog FXO
voice-port 1/0/0
input gain 10
no comfort-noise
connection plar 8000
caller-id enable
!
voice-port 1/0/1
input gain 10
no comfort-noise
connection plar 8000
caller-id enable
!
voice-port 1/1/0
input gain 10
no comfort-noise
connection plar opx 8898
!
voice-port 1/1/1
input gain 10
no comfort-noise
connection plar opx 8000
!
>>>> Dial-Out
!
dial-peer voice 3 pots
description Dialout via 3862
destination-pattern .
incoming called-number .
port 1/0/0
!
dial-peer voice 4 pots
description Dialout via 3015
preference 1
destination-pattern .
incoming called-number .
port 1/0/1
!
dial-peer voice 5 pots
description Dialout via 8797
preference 2
destination-pattern .
incoming called-number .
port 1/0/1
!
dial-peer voice 10 pots
description Dialout via 8898 for User-X <<<< Restriction Required
preference 3
destination-pattern .
incoming called-number .
port 1/1/0
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:10.10.10.1
!
>>> Incoming calls via POTS to SIP Server
dial-peer voice 2 voip
preference 1
destination-pattern 8...
progress_ind setup enable 3
session protocol sipv2
session target ipv4:10.10.10.1:5060
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
no vad
01-08-2012 02:08 AM
yes you can do that by configuring corelist, so that calls comming from 1000 will be assinged incoming corelist that has a memeber, and at the same time an outgoing corelist should be configured (which has that member) and should be assigned to the dial peer pointing to FXO port 0/0/0. so this will force extensiont 1000 to go through FXO port 0/0/0
you can do the same things to force the other extensions to go throught the other ports.
Best Regards,
Odai Jeriss
01-08-2012 08:49 AM
But the extensions are not local, this router is acting as SIP to PSTN and vice versa. Calls are landing in via a SIP trunk.
Thanks
-Azher
01-08-2012 09:50 AM
thats should not affect as you can configure an incomming dial peer which use "answer-address ANI_String" to catch the calling number in incomming calls and this dial peer will assign incomming corelist which will be used to direct the call to the specific FXO port.
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml
or you can use "incoming uri {called|calling} tag" see the following link for details
http://www.cisco.com/en/US/docs/ios/voice/ivr/configuration/guide/gt_url.html#wp1055648
Best Regards,
Odai Jeriss
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