Cisco CME & CUE - Incomming call transfer to external nunber not working

Unanswered Question
Jul 20th, 2009

Hope you are all well.

I have a problem with one of our client network. All VoIP calls and voice mail are working fine except they can't transfer/forward incoming call to external number on SIP trunk.

My client is using Cisco SME 4.0 and CUE 3.0. The error message is 'No Mactch Dial peer', when we transfer any call to mobile number. I have attached below translation rules and dial peers.

Could some help me to resolve this issue and let me know if you need any other information?




voice service voip

allow-connections sip to sip

redirect ip2ip


call start slow


registrar server expires max 3600 min 3600


voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw


voice register global

mode cme


max-dn 288

max-pool 96

dialplan-pattern 1 9000 extension-length 4



voice register pool 60

id mac 0013.FD42.1968

max registrations 96

codec g711ulaw


voice translation-rule 2

rule 1 /^999$/ /999/

rule 2 /^9(.*)/ /\1/


voice translation-rule 3

rule 1 /^.*/ /2033777820/



voice translation-profile PSTN_CallForwarding

translate redirect-target 3

translate redirect-called 3


voice translation-profile PSTN_Outgoing

translate calling 3

translate called 2

translate redirect-target 3

translate redirect-called 3


dial-peer voice 9000 voip

destination-pattern 900.


session protocol sipv2

session target ipv4:

dtmf-relay sip-notify

codec g711ulaw

no vad


dial-peer voice 200 voip

destination-pattern 2222

session protocol sipv2

session target ipv4:

dtmf-relay sip-notify


dial-peer voice 4 voip

description **Outgoing Call to SIP Trunk**

translation-profile outgoing PSTN_Outgoing

preference 7

destination-pattern 0T

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

no vad


dial-peer voice 5 voip

description **Incoming Call from SIP Trunk**

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

session target sip-server

incoming called-number .

dtmf-relay rtp-nte

no vad




no remote-party-id

retry invite 2

retry response 3

retry bye 2

retry register 10

retry options 0

timers connect 100

timers register 150






maximum bit-rate 386

pin 1234

load 7910 P00403020214

load 7960-7940 P0030702T023

load 7914 S00105000100

max-ephones 96

max-dn 96

ip source-address port 2000

auto assign 1 to 96

service phone videoCapability 1

timeouts interdigit 3

timeouts ringing 45

network-locale GB

time-format 24

date-format dd-mm-yy

voicemail 9000

max-conferences 12 gain -6

call-park system redirect

call-forward pattern .T




transfer-system full-consult dss

transfer-pattern .T

I have this problem too.
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sebastiaanv Thu, 08/13/2009 - 04:03

Hello Kash,

I too am in search of a solution for my CFWDALL CME problem (hairpin routing from & to PSTN).

May be this might help you with your SIP CFWD problem:

Check the rated reply at the bottom.


sebastiaanv Thu, 08/13/2009 - 04:26


I read your proposed solution for PSTN CFWD problem here:

The problem described by the OP is similar to my problem, so I'd thought to give your numbering type solution a go, or the loopback-dn solution, before I post my problem.

In this thread I was pointing out a possible solution for the original poster's problem.



Paolo Bevilacqua Thu, 08/13/2009 - 04:34

The loopback-dn is way too complicated to be of practical use, I had suggested that for a special requirement only.

There is also a nasty bug that can prevent CFW to work on ISDN with certain IOS versions.

In any case, if you have problems, take the debugs and post'em in a new thread.

sebastiaanv Thu, 08/13/2009 - 04:40

Thanks, I'll try the numbering-plan thing and if it fails I'll post a new thread with all the info. Hope it'll work, been working on it for hours now...


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