SIP incoming Dial-Peer not working we keeping hitting dialpeer 0

[yamikani2g2]

Good day Experts,

I followed the advise given and i configured the following to my suprise the inbound  calls cant even hit the dial peer voice 2 is there any thing done wrong here. attached are some debugs. show dialplan says it does not find a match.

Config extracts and attached are debug outputs.

dial-peer voice 2 voip
description SIP TRUNK TO PBX
session protocol sipv2
incoming called-number .
dtmf-relay rtp-nte
codec g711ulaw
no vad

**********************

voice translation-rule 8243
rule 1 /^0XXX367340$/ /8243113/

******************************************

voice translation-profile INBOUND
translate called 8243

****************************

dial-peer voice 2 voip
translation-profile incoming INBOUND

******************************************

dial-peer voice 4000 voip
description SIP TRUNK TO Yeaster
destination-pattern .
session protocol sipv2
session target ipv4:10.X.1.2XX
codec g711ulaw

[Sreekanth Narayanan]

This is happening because the incoming call arrives on the ISDN interface. This is not a VOIP interface. It is a POTS interface.

*Jan 13 21:01:01.366 UTC: %ISDN-6-DISCONNECT: Interface Serial0/2/0:21 disconnected from 977965787 , call lasted 14 seconds

Please change the dial-peer to "dial-peer voice 2 pots", or create a new dial-peer. This dial-peer will be matched after that.

[HARIS_HUSSAIN]

For Incoming Dial Peer Match the Dial-Peer Type should be same. IF call is coming on Analog/PRI lines then Pots dial-peer will match.
In you case as pointed by Sreekanth Narayanan you need to delete the current voip dialpeer and receate it as pots dialpeer as shown below.

dial-peer voice 2 pots
description SIP TRUNK TO PBX
session protocol sipv2
incoming called-number .
dtmf-relay rtp-nte
codec g711ulaw
no vad

[yamikani2g2]

Thanks Harris i will try that i can only test that over the weekend i will update the discussion board.

Could you point me somewhere where i can do some extensive reading and Labs on this technology will appreciate that so i get a deep understanding i feel need to refresh and be rock solid with the technology. much appreciated

[Maren Mahoney]

I would suggest not deleting the "dial-peer voice 2 voip" as it may be in use for other things (such as outbound dialing) but rather creating a new "dial-peer voice 3 pots" to capture the inbound pots call. If you want to test without the dial-peer voice 2 voip, you can go into the dial-peer and issue "shutdown" on it. This will cause the router to ignore the dial-peer.

[yamikani2g2]

Thank you experts i will do just that, Please share some resources i can use to solidify my Quest for knowledge.

[Maren Mahoney]

There is an ocean of documentation out there. Use Google to find what you are looking for at any particular moment.

Also, here is a link to Collaboration Training Videos that Cisco has published on a variety of topics. I have found the lengthy series of videos on SIP to be very educational!

Good luck in your studies. If you have more questions, we are here to help.

Collaboration Training Videos

Maren