Call forward to PSTN on cme

Question

Hi,
unable to set up call forward to PSTN.
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mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-qformat:yes;
mso-style-parent:"";
mso-padding-alt:0cm 5.4pt 0cm 5.4pt;
mso-para-margin:0cm;
mso-para-margin-bottom:.0001pt;
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font-size:11.0pt;
font-family:"Calibri","sans-serif";
mso-ascii-font-family:Calibri;
mso-ascii-theme-font:minor-latin;
mso-fareast-font-family:"Times New Roman";
mso-fareast-theme-font:minor-fareast;
mso-hansi-font-family:Calibri;
mso-hansi-theme-font:minor-latin;
mso-bidi-font-family:"Times New Roman";
mso-bidi-theme-font:minor-bidi;}
I have tried activating the Call forward via the phone or manually via the config, but when I attempt a call to IP Communicator from PSTN or via extn I am not seeing re-INVITE which should be generated for the forwarded call. Am i missing something?
    PSTN / IP phone ------> Calling extn on CME (which is call forwarded to another PSTN number)
config below:
voice service voip
ip address trusted list
  ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
  registrar server expires max 250 min 200
  asserted-id pai
  localhost dns:XXXXX
  outbound-proxy dns:XXXXX
dial-peer voice 100 voip
description ** Incoming call from SIP trunk **
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 1 
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
dial-peer voice 101 voip
description ** Outgoinging call to SIP trunk **
translation-profile outgoing SIPOUT
destination-pattern 1T
session protocol sipv2
session target sip-server
voice-class codec 1 
voice-class sip dtmf-relay force rtp-nte
voice-class sip profiles 101
dtmf-relay rtp-nte
no vad
dial-peer voice 102 voip
description ** Outgoinging call to SIP trunk **
destination-pattern 0[2-9].T
session protocol sipv2
session target sip-server
voice-class codec 1 
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
telephony-service
max-ephones 4
max-dn 12
ip source-address 192.168.100.2 port 2000
calling-number initiator
timeouts interdigit 5
load 7960-7940 P00308010200
date-format dd-mm-yy
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
transfer-system full-consult dss
transfer-pattern .T
transfer-pattern 0.T
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn  1  dual-line
number 4961 secondary 99474961 no-reg both
label 4961
name 4961
call-forward all 021605547

Answer 1

/* Style Definitions */
table.MsoNormalTable
{mso-style-name:"Table Normal";
mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-qformat:yes;
mso-style-parent:"";
mso-padding-alt:0cm 5.4pt 0cm 5.4pt;
mso-para-margin:0cm;
mso-para-margin-bottom:.0001pt;
mso-pagination:widow-orphan;
font-size:11.0pt;
font-family:"Calibri","sans-serif";
mso-ascii-font-family:Calibri;
mso-ascii-theme-font:minor-latin;
mso-fareast-font-family:"Times New Roman";
mso-fareast-theme-font:minor-fareast;
mso-hansi-font-family:Calibri;
mso-hansi-theme-font:minor-latin;}
Does a direct call (without forwarding) work through this dial-peer? YES
The session target of dial-peer 101 is the "sip-server". In wich way is configured? Is it an IP address or a name? FQDN
Can you ping it from the CME? YES
The CME can resolve the name via DNS? Resolved on the CME Can you post the sip-ua config?
sip-ua
credentials number 99474960 username 99474960 password 7 XXXXXXXXX realm as-test.xys.net 
authentication username 99474960 password 7 XXXXXXX 
calling-info pstn-to-sip asserted-id number set 99474960 
no remote-party-id 
disable-early-media 180 
retry invite 2
retry register 3
timers connect 100 
registrar dns:as-test.xys.net expires 60  sip-server dns:as-test.xys.net 
host-registrar