Cisco Voice SIP - Error 4404

[brian.vanoy]

We had some numbers relocated from a PRI circuit as VTNs on a SIP trunk and began to received Error 4404 at our outside interface when placing calls to the numbers.  We programmed the dial-peer (23) into the router, and have never had an issue with this before with any of the other numbers, including: 703, 304, 571.  

Can anyone see a problem in the config below.. The telco rep said our outside interface was rejecting the calls.

voice service voip
address-hiding
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol pass-through g711ulaw
h323
h225 display-ie ccm-compatible
modem passthrough nse codec g711ulaw
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
early-offer forced
midcall-signaling passthru
sip-profiles 100

voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8

voice class h323 1
  h225 timeout tcp establish 3

voice class sip-profiles 100
 request INVITE sip-header Allow-Header modify " UPDATE, " " "
 request REINVITE sip-header Allow-Header modify " UPDATE, " " "
 response 180 sip-header Allow-Header modify " UPDATE, " " "
 response 200 sip-header Allow-Header modify " UPDATE, " " "

voice translation-rule 1
 rule 1 /^9/ //
voice translation-profile outbound
 translate called 1
voice-card 0
 dsp services dspfarm

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

interface GigabitEthernet0/0
 description *** MPLS LINK ***
 ip address 64.x.x.x 255.255.255.252
 duplex full
 speed 100
 service-policy output MPLS-QOS

interface GigabitEthernet0/1
 ip address 192.168.x.x 255.255.255.0
 ip flow ingress
 duplex auto
 speed auto
 h323-gateway voip bind srcaddr 192.168.x.x

dial-peer voice 1 voip
 description *** Route Calls to CallManager ***
 preference 1
 destination-pattern [0-8]...
 voice-class codec 1
 voice-class h323 1
 session target ipv4:192.168.x.x
 incoming called-number ....
 dtmf-relay h245-alphanumeric
 ip qos dscp cs5 media
 no vad
!
dial-peer voice 2 voip
 description Outbound Local Calls to Paetec
 translation-profile outgoing outbound
 preference 2
 destination-pattern 9[2-9]......
 voice-class sip early-offer forced
 voice-class sip profiles 100
 session protocol sipv2
 session target ipv4:172.x.x.x
 dtmf-relay rtp-nte
 codec g711ulaw
!
dial-peer voice 3 voip
 description Outbound Long Distance Calls to Paetec
 translation-profile outgoing outbound
 preference 2
 destination-pattern 91[2-9].........
 voice-class sip early-offer forced
 voice-class sip profiles 100
 session protocol sipv2
 session target ipv4:172.x.x.x
 dtmf-relay rtp-nte
 codec g711ulaw
!
dial-peer voice 4 voip
 description Outbound International to Paetec
 translation-profile outgoing outbound
 preference 2
 destination-pattern 9011.T
 voice-class sip early-offer forced
 voice-class sip profiles 100
 session protocol sipv2
 session target ipv4:172.x.x.x
 dtmf-relay rtp-nte
 codec g711ulaw
!
dial-peer voice 5 voip
 description Outbound 911 Calls to Paetec
 translation-profile outgoing outbound
 preference 2
 destination-pattern 911
 voice-class sip early-offer forced
 voice-class sip profiles 100
 session protocol sipv2
 session target ipv4:172.x.x.x
 dtmf-relay rtp-nte
 codec g711ulaw
!
dial-peer voice 6 voip
 description Outbound 911 Calls to Paetec
 translation-profile outgoing outbound
 preference 2
 destination-pattern 9911
 voice-class sip early-offer forced
 voice-class sip profiles 100
 session protocol sipv2
 session target ipv4:172.x.x.x
 dtmf-relay rtp-nte
 codec g711ulaw
!
dial-peer voice 20 voip
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 incoming called-number 703.......
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 21 voip
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 incoming called-number 571.......
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 10 pots
 destination-pattern 7033650460
 port 0/0/0
!
dial-peer voice 100 voip
 description *** Route Calls to CallManager ***
 preference 2
 destination-pattern [0-8]...
 voice-class codec 1
 voice-class h323 1
 session target ipv4:192.168.x.x
 incoming called-number ....
 dtmf-relay h245-alphanumeric
 ip qos dscp cs5 media
 no vad
!
dial-peer voice 22 voip
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 incoming called-number 301.......
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 23 voip
 voice-class codec 1
 session protocol sipv2
 session target sip-server
 incoming called-number 904.......
 dtmf-relay rtp-nte
 no vad
!
!
sip-ua
 credentials username 571xxxxxxx password 7 xxxxxxxxxxx realm none
 no remote-party-id
 retry invite 2
 retry register 10
 timers connect 100
 registrar ipv4:172.x.x.x expires 3600
 sip-server ipv4:172.x.x.x
 host-registrar
!
!
!
call-manager-fallback
 max-conferences 4 gain -6
 transfer-system full-consult
 timeouts interdigit 5
 ip source-address 192.168.x.x port 2000
 max-ephones 30
 max-dn 144

[Jonathan Unger]

Hi There,

A couple things:

1. Can you please post the following debugs for a failed call? Be sure to disable console logging before running any of these. If it is a really busy gateway you can do these during a period of low activity.

- debug ccsip messages
- debug voip dialpeer inout
- debug voip ccapi inout

2. Are you trying to route these inbound calls to CUCM, or back out to your sip-ua defined "sip-server"? It isn't too clear based on your dial-peers where you would like these calls to go.

[brian.vanoy]

Jonathan,

Thank you for taking the time to respond to my inquiry. 

1. Unfortunately, we had to move the numbers back to the PRI temporarily to keep them operational, so a failed call with the debug I assume isn't an option at this point to gain any clarity.

2. Correct, they are inbound calls being routed to Call Manager, and I assumed like my other dialing peers, the "session target sip-server" would suffice for the routing once the numbers were allocated to the SIP trunk itself.

Thanks, Brian

[Jonathan Unger]

Hi Brian,

Cisco VGWs operate on the principal of inbound and outbound dial-peer matching, meaning that there must be an inbound dial-peer and an outbound dial-peer for each leg of the call.

In this case it looks like you have a valid inbound dial-peer (23), But no valid outbound dial-peer to CUCM.

Your other incoming DIDs in area codes 703, 304, and 571 are working almost by fluke because of dial-peer 1.

dial-peer voice 1 voip
description *** Route Calls to CallManager ***
preference 1
destination-pattern [0-8]...

The destination pattern states that the number must start with a number between 0 and 8 and have 3 or more digits after.

The other DIDs start with the number 7,3, and 5, so they will match that destination pattern and be routed to CUCM. If you wanted dial-peer 1 only to be good for 4 digits it would have to end with a $.

The new area code will not match dial-peer 1 because it starts with a 9 (not in range).

I would recommend creating a new CUCM dial-peer for the new set of DID's, here is an example:

dial-peer voice 999 voip
description *** Route Calls to CallManager ***
destination-pattern 904.......$
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.1.1
dtmf-relay h245-alphanumeric
ip qos dscp cs5 media
no vad

For more information on dial-peer matching on Cisco VGWs check out the following link:
http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/cube-dp.html

Please let us know if this helps!