URGENT dial-peer config

Unanswered Question

I have a problem will my SIP trunk. If I call from my cell phone into this trunk everything works and I see the proper dial-peers are being matched, but we have a client and they are sending us calls from their IVR to the same number, I am matching different dial-peers (default)which is really giving us some weird results.

My question is, if I have someone sending me "Restricted" in calling ( cisco-username=Restricted

----- ccCallInfo IE subfields -----

cisco-ani=Restricted) how can I get this call to match my dial-peer 100 instead of the default dial-peer. Here are the two dial-peers I want to match.

INcoming

dial-peer voice 100 voip

destination-pattern .T

session protocol sipv2

session target ipv4:xx.xx.xx.xx:4060

dtmf-relay rtp-nte digit-drop

codec g711ulaw

ip qos dscp cs5 media

ip qos dscp cs5 signaling

no vad

And outgoing

dial-peer voice 208 voip

preference 1

destination-pattern 30[7-8][0-9]

voice-class h323 1

session target ipv4:172.16.1.4

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

But like I said it is matching the default dial-peer when someone is sending me restricted.

I have this problem too.
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Paolo Bevilacqua Fri, 06/27/2008 - 12:57

Hi,

To DP 100 add: "incoming called-number .".

Then, you don't need destination-pattern there.

This should make DP 100 a valid incoming for any voip call, regardless if calling is restricted or not.

Please rate post if it does!

Thank you that gets me to match the correct dial-peer, now let me tell you the root problem.

If I call from my cell phone I get the results I expect, but if the call is transferred from this IVR the dial-peers now match corrects, but I only hear part of the greeting and it does not notice my dtmf???? How can it only play a small part of the greeting?

Paolo Bevilacqua Fri, 06/27/2008 - 13:13

Hi, any chance you've packet loss somewhere ? If you call from a fixed phone, can you hear the prompts ok ?

For the tones, try dtmf-relay cisco-rtp on both DPs. You also have digit-drop, I would remove that as well.

Paolo Bevilacqua Fri, 06/27/2008 - 13:36

No, you can do all testing needed. Once you realize what is he condition that causes poor audio, mention it here.

I just wanted to reply and let everyone know my findings

After debugging my IVR script I noticed that when I called from our clients IVR they are adding 159 which will take you to different places within the script other then the main greeting, which is why we never heard the main greeting.

p.bevilacqua thank you for all of your help.

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