incoming calls "network busy"

[Ansik]

So i had incoming calls work before with a test number received from the provider. Today they changed the number on their end to what it's suppose to be and  incoming calls are now failing with message 500 internal server error and cause 16. Why is the call failing?

Call flow is like this PSTN==>CME==>IP phone

dial-peer voice 200 voip
 description ** Incoming dialpeer sip-trunk calls **
 Destination-pattern 21
 session protocol sipv2
 session target sip-server
 session transport udp
 incoming called-number .
 dtmf-relay rtp-nte
 codec g711alaw
 no vad

[M02@rt37]

Hi @Ansik

If this can help you about the error 500 internal server.

http://docwiki.cisco.com/wiki/SIP_Troubleshooting:_SIP_Calls_Receives_500_Internal_Server_Error_"Routing_Failed"_Event

[gmgarrian]

The call is looping back to the PSTN.

It's hitting dial peer 2 on the inbound leg and hitting dial peer 3 for the outbound leg.

Can you upload the outputs of:

show version

show config

show dial-peer voice summary

Thanks!

[Ansik]

@gmgarrian

attached outputs

[gmgarrian]

Hi Ansik,

The inbound call has a DNIS of 475994:

Received: 
INVITE sip:475994@10.114.11.18:5060;user=phone SIP/2.0

 That matches dial peer 2 because of the "incoming called-number." here:

dial-peer voice 2 voip
destination-pattern 21
session protocol sipv2
session target sip-server
session transport udp
incoming called-number .
dtmf-relay rtp-nte
no vad

Because there is no translation rule it will hunt for a destination pattern that matches the DNIS of 475994.

The only dial peer that will match a six digit number is dial peer 3:

dial-peer voice 3 voip
description ** Outgoing dialpeer Landline Calls **
destination-pattern [2-6].....$
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad

Since your phones all have two digit extensions you will need to add a translation rule to the incoming peer to translate the DNIS to the desired ephone dn number.

Which phone should this call route to?  Even if you strip the DNIS to the last two digits you get 94 which does not match any IP phone's DN.

[Ansik]

I had a translation profile configured on the dial-peer 2 as per below but calls were still failing so that's why i tried to add a destination-pattern directly to extension 21.
voice translation-rule 2
rule 1 /47..../ /21/
!
voice translation-profile INCOMING-CALLS
translate called 2

[gmgarrian]

That translation rule should work assuming it was applied on the dial peer correctly.  But, now that you pointed out the destination number is 21 you have an overlapping dial plan.  The virtual dial-peer 20005 and the configured peer 2 both have destination patterns of 21.  Peer 2 routes it back to the PSTN.

Can you re-apply the translation rule to peer 2 with this command and remove the destination pattern:

dial-peer 2 voice 2 voip

translation-profile incoming INCOMING-CALLS

no destination-pattern 21

Then, make a test call and get the following debugs:

debug ccsip message

debug voip ccapi inout

debug sccp events

debug sccp packets

Thanks!

[Ansik]

will apply and test again. Just got notified that the provider switched the number back to analog so will have to wait until they put the number back on the sip trunk.

You are right it was overlapping but oddly enough i had no problems before.

[Ansik]

attached the log.

i have added the translation rule to the dial-peer . the call is still failing

dial-peer voice 2 voip
 description ** Incoming dialpeer sip-trunk calls **
 translation-profile incoming INCOMING-CALLS
 session protocol sipv2
 session target sip-server
 session transport udp
 incoming called-number .
 dtmf-relay rtp-nte
 codec g711alaw
 no vad

!

voice translation-rule 2
 rule 1 /47..../ /21/

!

voice translation-profile INCOMING-CALLS
 translate called 2

Also i modified dial-peer 3 so it doesnt overlap anymore to

dial-peer voice 3 voip
 description ** Outgoing dialpeer Landline Calls **
 destination-pattern 0[2-6].....$
 session protocol sipv2
 session target sip-server
 session transport udp
 dtmf-relay rtp-nte
 codec g711alaw
 no vad