incoming dial-peer syntax

[ccg-collab1]

Hi All,

Would like to understand the syntax.

If there is incoming call 995XXX should it be match to this incoming dial-peer.

dial-peer voice 200 voip

destination-pattern .T

session target ipv4: cucm ip

incoming called-number #

tmf-relay h245-signal h245-alphanumeric
codec g729r8 bytes 40
fax-relay ecm disable
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
ip qos dscp cs5 media

Thank you

[Manish Gogna]

Is this the only dial-peer configured on the router?

Why have you configured "incoming called-number #" , configure "incoming called-number ." instead.

Rest looks okay.

HTH

Manish

[Rajan]

Hi,

As Manish mentioned, it also depends on the other dial-peers configured to see whether this dial-peer will be used as an incoming dial-peer. Also let us know the call flow to confirm that.

HTH

Rajan

[ccg-collab1]

Hi Rajan,

Here is the call flow.

SIP Phone >>> CUCM >>> H323_TMP >>> H323_TGN >>> WAN >>> PABX >>> IP Phone 

Actually I originally asked for what is the meaning of syntax # in the incoming called-number as we are encountering call drop in incoming call from H323 TGN going to H323 TMP maybe this is related to the problem. Let me know if you will need the sho run of the both router so you could check the config if its correct. Thank you

dial peer configured on H323_TGN

dial-peer voice 200 voip
destination-pattern .T
session target ipv4:183.81.244.172
incoming called-number #
dtmf-relay h245-signal h245-alphanumeric
codec g729r8 bytes 40
fax-relay ecm disable
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
ip qos dscp cs5 media
clid network-number 8811063
!
dial-peer voice 201 voip
description ### TO TMP CUCM ###
preference 1
destination-pattern 965[012]T
session protocol sipv2
session target ipv4:10.1.4.4
dtmf-relay h245-signal h245-alphanumeric
codec g729r8 bytes 40
fax-relay ecm disable
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
ip qos dscp cs5 media
!
dial-peer voice 202 voip
description ### TO TMP CUCM ###
preference 2
destination-pattern 965[012]T
session protocol sipv2
session target ipv4:10.1.66.4
dtmf-relay h245-signal h245-alphanumeric
codec g729r8 bytes 40
fax-relay ecm disable
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
ip qos dscp cs5 media

================================================================

and the dial-peer configured from H323 TMP

dial-peer voice 8000 voip
description *** To TGN Network C3945 ***
destination-pattern .T
session target ipv4:202.160.196.43
dtmf-relay h245-signal h245-alphanumeric
codec g729r8 bytes 40
fax-relay ecm disable
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
no vad
!
dial-peer voice 8001 voip
description *** To TMP CUCM ***
preference 1
destination-pattern 965[012]T
session target ipv4:10.1.4.4
dtmf-relay h245-signal h245-alphanumeric
codec g729r8 bytes 40
fax-relay ecm disable
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
no vad
!
dial-peer voice 8002 voip
description *** To TMP CUCM ***
preference 2
destination-pattern 965[012]T
session target ipv4:10.1.66.4
dtmf-relay h245-signal h245-alphanumeric
codec g729r8 bytes 40
fax-relay ecm disable
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
no vad
!
dial-peer voice 8003 voip
description *** Incoming Call ***
answer-address .
voice-class codec 1

Hope for your response.

[Rajan]

BTW what are these gateways H323 TMP & H323 TGN used for. You have given dial-peers to same CUCM servers from both these gateways. So bit confused about the call flow here.

[ccg-collab1]

Hy Rajan,

Call Flow

IP Phone 1 >>> TMP CUCM >>> TMP H323 GW >>> TGN H323 GW >>> PABX >>> IP Phone 2

The TMP H323 GW is registered on the TMP CUCM and the TGN H323 GW is not registered on any PABX so basically, when IP Phone 1 calling to IP Phone 2, the call will pass thru the TMP H323 Gateway first then the call will pass to TGN H323 Gateway going to IP Phone 2. 

We can call from IP Phone 1 to IP Phone 2 successfully however when IP Phone 2 is calling to IP Phone 1 the call is just ringing after picking up, we received Cause Code 47 or resource unavailable. We are negotiating via G729. Is there something missing on my configuration based on the dial peers configured on both Gateway that I showed you?

Appreciate your help.

[Rajan]

ah..ok..

Then do you have a trunk to H323_TGN. The reason I ask is you have a dial-peer directly from this gateway to CUCM. I suppose from what I understood, the call from IP Phone 2 to IP Phone 1 should go thorugh  TGN H323 GW and TMP H323 GW. Correct me if am wrong.

and where you see reason code 47. I mean on which gateway. Can you share the debugs with the call details to check. If one side is H323 and other one is SIP, then we need both the below debugs:

debug ccsip messages

debug voip ccapi inout

Thanks 

Rajan

[Pulkit Sharma]

Hi CCG-collab,

The "incoming called-number #" which you have configured would never get triggered for the configuration you have shared. This might if you have any kind of translation set for the called number which translate the DNIS to say #995XXX, but I do not see one applied on the dial-peer.
Owing to which the voice-gateway would look at the DNS and since it would not find "#" in the dialed string, it would use the next parameter for the incoming dial-peer selection. In your case it would use the "destination-pattern .T".

Once the incoming dial-peer is selected the router will again look into the DNIS , however this time it would only use the destination-pattern to route the call.

Here is the order for the selection for the incoming dial-peer:

1)Incoming called-number  ---> Matches DNIS
2)Answer-address              ---> Matches ANI
3)destination-pattern          ---> Matches ANI
4)voice-port
5)If no match is found in the first four steps, then the default dial peer 0 would be selected.

Note: You should ideally not want dial-peer 0 to selected, as it restricts negotiation of certain parameter for  VOIP call and might cause anomalous behavior. So try to avoid it.

I hope it builds a better perspective in dealing with your problem.

You can refer the following link:

http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/14074-in-dial-peer-match.html

Thanks
Pulkit