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Restricted Pots Dial for a specific calling number

azher.amin
Level 1
Level 1

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

3 Replies 3

odaijeriss
Level 1
Level 1

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

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

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