Seeking assistance for CUBE ENUM config

bill.efthimiou · ‎08-14-2012

Hi all,

I am experimenting in our lab with using ENUM lookup and a CUBE (3845 with IOS 15.1(3)T3).

I have followed the document posted on the Cisco example configuration website: http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a0080ad7b94.shtml

Everything works as per the document. That is, when I run the test enum command, the result is as illustrated in the doc.

However, the call is NOT actually extended using the replacement URI. That is the call is never placed to the destination that corresponds to the resulting ENUM query. Instead we simply respond with a 404 Not Found SIP message.

I have a feeling I am missing something but which I can't seem to identify.

I have never had this working.

The ENUM server is running on a Ubuntu linux server using bind9.

Relevant IOS config is as follows;

ip domain name <our domain>

ip name-server 202.158.213.195

!

voice call send-alert

voice call disc-pi-off

voice call convert-discpi-to-prog

voice rtp send-recv

!

voice service voip

ip address trusted list

ipv4 182.255.112.18 255.255.255.255

no ip address trusted authenticate

address-hiding

allow-connections sip to sip

signaling forward unconditional

sip

session transport tcp

header-passing

error-passthru

early-offer forced

no call service stop

!

voice translation-rule 1

rule 1 /^02$.*$/ /612\1/

!

voice translation-profile Calls_via_APAN

translate called 1

!

voice enum-match-table 1

rule 1 1 /^61$.*$/ /+61\1/ e164.test

rule 2 1 /^02$.*$/ /+612\1/ e164.test

!

voice enum-match-table 2

rule 1 1 /(.*)/ /\1/ e164.test

!

interface <snip>

description LAB LAN

ip address 202.158.213.203 <snip>

!

dial-peer voice 1 voip

description incoming dial-peer matching

session protocol sipv2

session transport tcp

incoming called-number .

voice-class sip early-offer forced

dtmf-relay rtp-nte

codec transparent

!

dial-peer voice 10000 voip

destination-pattern 0297796908

session protocol sipv2

session target enum:1

dtmf-relay rtp-nte

codec transparent

!

dial-peer voice 10001 voip

destination-pattern 61297796908

session protocol sipv2

session target enum:2

dtmf-relay rtp-nte

codec transparent

!

dial-peer voice 10002 voip

destination-pattern 33333

session protocol sipv2

session target ipv4:182.255.112.16

incoming called-number 0297796908

dtmf-relay rtp-nte

codec transparent

!

-------------------------------------------------------------------------------------------

IOSGK_lab#term mon
IOSGK_lab#debug voice en
IOSGK_lab#debug voice enum det
IOSGK_lab#debug voice enum detail
enum debug detail debugging is on
IOSGK_lab#
IOSGK_lab#test en
IOSGK_lab#test enum 1 0297796908
IOSGK_lab#enum_resolve_domain: match_num 0297796908 table_indx 1
enum_resolve_domain: rule 2 result string +61297796908
generate_enum_search_string : search string 8.0.9.6.9.7.7.9.2.1.6.e164.test
enum_dns_query: name = 8.0.9.6.9.7.7.9.2.1.6.e164.test type = 35, ns_server = 0
order 100 pref 10 service SIP+E2U flag u
regexp !^.*$!sip:61262112650@182.255.112.18! replacement
num_elem = 1
NAPTR Record : order 100 pref 10 service SIP+E2U
               flags u regexp !^.*$!sip:61262112650@182.255.112.18!
               replacement
decode_naptr_record : re_string ^.*$
decode_naptr_record : re_substitution_string sip:61262112650@182.255.112.18
decode_naptr_record : re_flags_string
U_FLAG case, stopping query
new_e164_user sip:61262112650@182.255.112.18
contact_list :
              sip:61262112650@182.255.112.18
enum_resolve_domain: contact_list 69F802AC
contact_list :
              sip:61262112650@182.255.112.18
enum_test_command: contact_list 69F802AC

IOSGK_lab#
IOSGK_lab#test en
IOSGK_lab#test enum 2 61297796908
IOSGK_lab#enum_resolve_domain: match_num 61297796908 table_indx 2
enum_resolve_domain: rule 1 result string 61297796908
generate_enum_search_string : search string 8.0.9.6.9.7.7.9.2.1.6.e164.test
enum_dns_query: name = 8.0.9.6.9.7.7.9.2.1.6.e164.test type = 35, ns_server = 0
order 100 pref 10 service SIP+E2U flag u
regexp !^.*$!sip:61262112650@182.255.112.18! replacement
num_elem = 1
NAPTR Record : order 100 pref 10 service SIP+E2U
               flags u regexp !^.*$!sip:61262112650@182.255.112.18!
               replacement
decode_naptr_record : re_string ^.*$
decode_naptr_record : re_substitution_string sip:61262112650@182.255.112.18
decode_naptr_record : re_flags_string
U_FLAG case, stopping query
new_e164_user sip:61262112650@182.255.112.18
contact_list :
              sip:61262112650@182.255.112.18
enum_resolve_domain: contact_list 69F7FE84
contact_list :
              sip:61262112650@182.255.112.18
enum_test_command: contact_list 69F7FE84

-------------------------------------------------------------------------------------------------------------------------------------------------------

IOSGK_lab#debug ccsip mess

IOSGK_lab#debug ccsip messages

SIP Call messages tracing is enabled

IOSGK_lab#

*Aug 15 2012 13:20:52.943 AEST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0297796908@202.158.213.203:5060 SIP/2.0
Via: SIP/2.0/TCP 182.255.112.16:5060;branch=z9hG4bK5jtvut008gi10jot70g0.1
Call-ID: c1d35800-2b119d9-17955-23cf9eca@202.158.207.35
CSeq: 101 INVITE
Remote-Party-ID: "Bill Efthimiou" <sip:6952@202.158.207.35>;party=calling;screen=yes;privacy=off
Contact: <sip:6952@182.255.112.16:5060;transport=tcp>;video;audio;video
From: "Bill Efthimiou" <sip:6952@182.255.112.16:28954>;tag=1014027~968f888f-ba3a-4701-bd83-3fce8ea74ec4-33963081
To: <sip:+61297796908@202.158.213.203:5060>
Max-Forwards: 14
Allow: INVITE,OPTIONS,INFO,BYE,CANCEL,ACK,PRACK,REFER,SUBSCRIBE,NOTIFY
User-Agent: Cisco-CUCM8.6
Expires: 180
Date: Wed, 15 Aug 2012 03:39:05 GMT
Supported: timer,resource-priority,replaces,X-cisco-srtp-fallback,Geolocation
Session-Expires: 1800
Min-SE: 1800
Allow-Events: presence,kpml
X-TAATag: c3862244-e68a-11e1-a613-0010f316ecc8
Call-Info: <sip:202.158.207.35:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 3251853312-0000065536-0000004605-0600809162
Content-Length: 0

*Aug 15 2012 13:20:52.951 AEST: //1295/C1D358000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 182.255.112.16:5060;branch=z9hG4bK5jtvut008gi10jot70g0.1
From: "Bill Efthimiou" <sip:6952@182.255.112.16:28954>;tag=1014027~968f888f-ba3a-4701-bd83-3fce8ea74ec4-33963081
To: <sip:+61297796908@202.158.213.203:5060>
Date: Wed, 15 Aug 2012 03:20:52 GMT
Call-ID: c1d35800-2b119d9-17955-23cf9eca@202.158.207.35
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

*Aug 15 2012 13:20:52.955 AEST: //1295/C1D358000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 182.255.112.16:5060;branch=z9hG4bK5jtvut008gi10jot70g0.1
From: "Bill Efthimiou" <sip:6952@182.255.112.16:28954>;tag=1014027~968f888f-ba3a-4701-bd83-3fce8ea74ec4-33963081
To: <sip:+61297796908@202.158.213.203:5060>;tag=A15BC8-2113
Date: Wed, 15 Aug 2012 03:20:52 GMT
Call-ID: c1d35800-2b119d9-17955-23cf9eca@202.158.207.35
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=3
Content-Length: 0

*Aug 15 2012 13:20:52.959 AEST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:0297796908@202.158.213.203:5060 SIP/2.0
Via: SIP/2.0/TCP 182.255.112.16:5060;branch=z9hG4bK5jtvut008gi10jot70g0.1
CSeq: 101 ACK
Call-ID: c1d35800-2b119d9-17955-23cf9eca@202.158.207.35
From: "Bill Efthimiou" <sip:6952@182.255.112.16:28954>;tag=1014027~968f888f-ba3a-4701-bd83-3fce8ea74ec4-33963081
To: <sip:+61297796908@202.158.213.203:5060>;tag=A15BC8-2113
Max-Forwards: 14
Content-Length: 0

Any guidance here would be greatly appreciated. Please let me know if any further details are required to diagnose this problem.

Regards,

Bill

Karthik Sivaram · ‎08-19-2012

hi Bill,

cause of 3 ...indicates that the called party can't be reached . Are you able to ping the destination?

Could you include the following for me...

debug voip ccapi inout

debug ccsip all

(collect the above outputs in a buffer and copy/ upload the "sh log" in a clear text file to this post)

Thanks,

Karthik

bill.efthimiou · ‎08-19-2012

Thanks Karthik,

I have uploaded the debug to the original post. I am not sure I'm reading the debug correctly, but looks like a possible codec issue:

*Aug 20 2012 11:26:45.797 AEST: //-1/xxxxxxxxxxxx/SIP/Error/voipCodec_to_rtpAvpCodec: Unexpected VoIPCodec Type :No Codec

*Aug 20 2012 11:26:45.797 AEST: //22221/9D9BC0800000/SIP/Error/sipSPIAddSDPMediaPayload: Call Origination Failed: None of the selected codec from CLI is supported by SIP

*Aug 20 2012 11:26:45.797 AEST: //22221/9D9BC0800000/SIP/Error/sipSPIOutgoingCallSDP: Error with codec types on media line : 1

*Aug 20 2012 11:26:45.797 AEST: //22221/9D9BC0800000/SIP/Error/sipSPICreateOutboundSDP: Error in creating an SDP for the outbound call - Check for supported codecs

*Aug 20 2012 11:26:45.797 AEST: //22221/9D9BC0800000/SIP/Error/ccsipDoPostDNSAndENUMProcForInitialInvite: Error during deferred outbound SDP creation

*Aug 20 2012 11:26:45.797 AEST: //22221/9D9BC0800000/SIP/Error/sipSPIResolveFromContactList: ccsipDoPostDNSAndENUMProcForInitialInvite failed

Kind Regards,

Bill Efthimiou

Karthik Sivaram · ‎08-19-2012

hi Bill,

From the logs The cube has with reponded "404 Not found" cause =3 as ENUM query fails or not

resolved.

Hence the ENUM part of the config needs to be checked. I noticed some of the config is different from what is given in the doc . For eg. you have stated e.164.test when it should have been stated as e.164.arpa . Also check the DNS config.

Also on the CUCM set the "Stop Routing on Unallocated Number Flag to False" this will allow calls to completed even if ENUM query fails.

Hope this Helps!

Thanks,

Karthik