07-20-2009 07:33 AM - edited 03-15-2019 07:02 PM
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?
Thanks,
Kash
---------------------------------------------------------
voice service voip
allow-connections sip to sip
redirect ip2ip
h323
call start slow
sip
registrar server expires max 3600 min 3600
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
!
voice register global
mode cme
source-address xxx.xxx.xxxx.xxx
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.
b2bua
session protocol sipv2
session target ipv4:192.168.18.251
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 200 voip
destination-pattern 2222
session protocol sipv2
session target ipv4:192.168.17.37
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
!
!
sip-ua
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
sip-server ipv4:xxx.xxx.xxx.xxx
host-registrar
!
telephony-service
video
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 192.168.17.250 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
moh hotelcostes4.au
dn-webedit
time-webedit
transfer-system full-consult dss
transfer-pattern .T
08-13-2009 04:03 AM
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.
--bas
08-13-2009 04:06 AM
What is your problem ?
Please post new thread for new problems.
08-13-2009 04:26 AM
Hi,
I read your proposed solution for PSTN CFWD problem here: http://forums.cisco.com/eforum/servlet/NetProf?page=netprof&forum=Unified%20Communications%20and%20Video&topic=IP%20Telephony&topicID=.ee6c829&CommCmd=MB%3Fcmd%3Dpass_through%26location%3Doutline%40%5E1%40%40.1dded34b/6#selected_message
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.
Regards,
--bas
08-13-2009 04:34 AM
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.
08-13-2009 04:40 AM
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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