Choose a port to Outbound Dial Peer

[robson.mcts]

Hi everyone,

I would like to know if is possible choose a port for Outbound Call based in ANI Number.

By exemple:

I have a gateway ISR with 2 ports E1

E1-1 Prefix 7000-7900

E1-2 Prefix 4000-4999

This gateways have a SIP Trunk with a Lync Server.

The Lync do a call to PSTN

Calling From 7001 to PSTN Number 21341234 -> I would that this outbound call left by E1-1 and

Calling From 4001 to PSTN Number 21341234 -> I would that this outbound call left by E1-2

I follow this DOC and the translation do not work

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a00801bc341.shtml

On Debug of Translte the output is

Jul 25 23:30:55.908: xrule_checking calling 4001, called +90217193952442

Jul 25 23:30:55.908: xrule_checking peer_tag 2, direction 1, protocol 3

Jul 25 23:30:55.908: h323_xrule

Jul 25 23:30:55.908: h323_xrule

Jul 25 23:30:55.908: xrule_checking Return rc = -4

Jul 25 23:30:55.908: xrule_checking

Jul 25 23:30:55.908: xrule_checking calling 4001, called 90217193952442

Jul 25 23:30:55.908: xrule_checking peer_tag 6, direction 2, protocol 3

Jul 25 23:30:55.908: xrule_checking Return rc = -5

dial-peer voice 4 pots

destination-pattern .....T

progress_ind setup enable 3

direct-inward-dial

port 0/0/0:0

!

dial-peer voice 6 pots

destination-pattern ........T

progress_ind setup enable 3

direct-inward-dial

port 0/1/0:0

!

dial-peer voice 100 voip

shutdown

destination-pattern ....

rtp payload-type comfort-noise 13

session protocol sipv2

session target ipv4:10.145.68.5

session transport tcp

voice-class sip block 183 sdp present

dtmf-relay sip-kpml rtp-nte sip-notify

codec g711alaw

fax protocol none

ip qos dscp cs4 signaling

!

dial-peer voice 1 voip

translation-profile incoming 7300

answer-address 7496

rtp payload-type comfort-noise 13

session protocol sipv2

session target ipv4:10.145.68.5

session transport tcp

voice-class sip block 183 sdp present

dtmf-relay sip-kpml rtp-nte sip-notify

codec g711alaw

fax protocol none

ip qos dscp cs4 signaling

!

dial-peer voice 2 voip

translation-profile incoming 4000

answer-address 4001

rtp payload-type comfort-noise 13

session protocol sipv2

session target ipv4:10.145.68.5

session transport tcp

voice-class sip block 183 sdp present

dtmf-relay sip-kpml rtp-nte sip-notify

codec g711alaw

fax protocol none

ip qos dscp cs4 signaling

!

dial-peer voice 3299 pots

destination-pattern 3299T

port 0/0/0:0

!

dial-peer voice 2134 pots

destination-pattern 2134T

port 0/1/0:0

dial-peer voice 1 voip

translation-profile incoming 7300

answer-address 7496

dial-peer voice 2 voip

translation-profile incoming 4000

answer-address 4001

voice translation-profile 7300

translate called 1

!

voice translation-profile 4000

translate called 2

voice translation-rule 1

rule 1 /^9/  /3299/

!

voice translation-rule 2

rule 1 /^9/ /2134/

dial-peer voice 3299 pots

destination-pattern 3299T

port 0/0/0:0

dial-peer voice 2134 pots

destination-pattern 2134T

port 0/1/0:0

[paolo bevilacqua]

Have the other system prefix steering digits to called number.

[robson.mcts]

Hi Paolo,

Thanks by reply!

What can I delete?

Im learn dial-peer still.. sorry

[j.huizinga]

What Paola is saying is that Lync should do a little more.

When extension 4001 whats to make an external call, lync should prefix this with '11' for example

And when extension 7001 makes an external call, lync should prefix this with '12' (again example)

On the ISR you make two dial-peers; one for 11T and one for 12T

The dial peer 11T point to the  first E1, and 12T to the second.

JH

[robson.mcts]

Understand j.huizinga

But lync can not do it ...

Change the Called number based on Calling number

It should be manipulated in the ISR

This is possible ?

[j.huizinga]

Don't know Lync that well, but my colleague knows Lync very well and he has done that.

Problably two voice policies and apply this to the correct users

JH

[robson.mcts]

yeah dear..

But we have 1000 users and is needed create a policy call to separate these users based your number.... 

I'm looking for a way to make it easier on the gw.

[j.huizinga]

Then play with answer address, might work

JH

[robson.mcts]

Cool...

Can you give me a light ?

Im with this problem of thread

[j.huizinga]

two pots dial-peers each to a E1 line and with the same destination.

dial-peer voice 1 pots

answer-address 4...

destination-pattern

port 0/0/0

dial-peer voice 2 pots

answer-address 7...

destination-pattern

port 0/0/1

Don't know if this works, normally I make these decisions on Callmanager, or in your case Lync

JH

[robson.mcts]

JH dont work :/

The call always left by one dial-peer

[robson.mcts]

Dear

Lacked a configuration highlighted below and the Lync sent + before 9. Solved send 9 without + and all work like expected!

dial-peer voice 3299 pots
destination-pattern 3299T
translate-outgoing called 1
port 0/0/0:0

dial-peer voice 2134 pots
destination-pattern 2134T
translate-outgoing called 2
port 0/1/0:0