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771
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5
Helpful
17
Replies

dial-peer for specific phone...

snickered
Level 1
Level 1

I am trying to have a single extension use a particular inbound dial-peer. I figured I could do this by adding a dial-peer that is the same as the current dial-peer and add "answer-address" along with it. But that didn't work. Here's what I have:

dial-peer voice 100 voip

description ****INCOMING 10-DIGIT****

incoming called-number 9[2-9]..[2-9]......

voice-class codec 1

session protocol sipv2

dtmf-relay rtp-nte

no vad

!

dial-peer voice 500 voip

description ****INCOMING (SPECIAL)****

translation-profile incoming some-translation

incoming called-number 9[2-9]..[2-9]......

answer-address 5001

voice-class codec 1

session protocol sipv2

dtmf-relay rtp-nte

no vad

How do I accomplish what I'm trying to do? TIA.

1 Accepted Solution

Accepted Solutions

I didn't hear of so many difficulties in matching by calling number.

You can try updating IOS to see if behaves differently, or collapse voip DP into one. That does not interfere with pots ones.

View solution in original post

17 Replies 17

paolo bevilacqua
Hall of Fame
Hall of Fame

Lower preference on DP 100.

I figured lowering the preference on DP 100 would just make everything match DP 500. Based on http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml#topic3 my understanding is that once "incoming called-number" is matched the DP is used.

So, will lowering the preference force "answer-address" to be checked?

Yes, it should.

Hmmm... that didn't work for me. I don't think preference is working for me at all. When I have:

dial-peer voice 100 voip

description ****INCOMING 10-DIGIT****

preference 2

incoming called-number 9[2-9]..[2-9]......

voice-class codec 1

session protocol sipv2

dtmf-relay rtp-nte

no vad

dial-peer voice 500 voip

description ****INCOMING (SPECIAL)****

preference 1

incoming called-number 9[2-9]..[2-9]......

voice-class codec 1

session protocol sipv2

dtmf-relay rtp-nte

no vad

It still doesn't use DP 500. How is the preference supposed to work?

Right, might not even work for incoming DP. Try no huntstop on DP 100.

That didn't work either. I think 'no huntstop' is the default because it didn't show in the running-config. I tried with and without huntstop on each DP with no success. Any other ideas?

Can you try removing incoming called-number and leave answer-address only ?

If I remove the incoming called-number on both then other people won't be able to make calls. Is that what you mean?

Remove it on DP 500 and answer-address should cause a match.

When I do that 'debug dialpeer' doesn't show it as a match at all. The order of operations is that it checks all dialpeers for 'incoming called-number' and if none are matched it will move on to 'answer-address'. So, 'answer-address' will never be matched if I have 'incoming called-number' on any other dialpeer that matches. Here's what I tried:

dial-peer voice 100 voip

description ****INCOMING 10-DIGIT****

incoming called-number 9[2-9]..[2-9]......

voice-class codec 1

session protocol sipv2

dtmf-relay rtp-nte

no vad

!

dial-peer voice 500 voip

description ****INCOMING (SPECIAL)****

answer-address 5001

voice-class codec 1

session protocol sipv2

dtmf-relay rtp-nte

no vad

Then, can you try removing incoming called-number from all and use answer-address . on DP 100 ?

I have several other dial-peers that rely on 'incoming called-number'. I can't really remove that from all dial-peers and call it a "fix". Surely there is a way to force a single phone to go one way and all other phones go another. Seems like it would be simpler than this.

What exact IOS are you running ?

What differentiates your incoming voip DPs that prevents you from grouping them ?

I am running 12.3(26) on a AS5300. I was thinking about all my inbound dial-peers (pots included). But you said voip and now I that I'm only considering my inbound voip dial-peers it might be possible. Here are my voip dial-peers:

dial-peer voice 100 voip

incoming called-number 9[2-9]..[2-9]......

voice-class codec 1

session protocol sipv2

dtmf-relay rtp-nte

no vad

!

dial-peer voice 101 voip

incoming called-number 91[2-9]..[2-9]......

voice-class codec 1

session protocol sipv2

dtmf-relay rtp-nte

no vad

!

dial-peer voice 102 voip

incoming called-number 9011T

voice-class codec 1

session protocol sipv2

dtmf-relay rtp-nte

no vad

I guess I could do something like:

dial-peer voice 100 voip

answer-address ....

voice-class codec 1

session protocol sipv2

dtmf-relay rtp-nte

no vad

!

dial-peer voice 500 voip

answer-address 5001

voice-class codec 1

session protocol sipv2

dtmf-relay rtp-nte

no vad

Do people do this? A.K.A. Is this good practice?

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