Hi Mohsin,
As the cause value suggest 484 Address incomplete is seen when called number routing is not configured at ITSP/Telco end. Can you verify if called /calling number routing is configured for the digits we are sending to them?
Also verify what cause value does PSTN see at their end? Is it 47? if so then verify codec configured at their end?
Incase it is different then we need to configure Xcoder/mtp?
Snippet from logs:
=============
INVITE sip:00501029946@10.1.0.4:5060 SIP/2.0
Via: SIP/2.0/TCP 172.16.200.20:5060;branch=z9hG4bK24638b7f7e7c
From: "Test11" <sip:4945001@172.16.200.20>;tag=173064~7a4acf75-5f09-4773-8466-a6fba898a872-46739733
To: <sip:00501029946@10.1.0.4>
Date: Tue, 01 Sep 2015 19:42:50 GMT
Call-ID: 9f901e80-5e51ffba-14df9-14c810ac@172.16.200.20
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:172.16.200.20:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 2677022336-0000065536-0000085445-0348655788
Session-Expires: 1800
P-Asserted-Identity: "Test11" <sip:4945001@172.16.200.20>
Remote-Party-ID: "Test11" <sip:4945001@172.16.200.20>;party=calling;screen=yes;privacy=off
Contact: <sip:4945001@172.16.200.20:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 423
v=0
o=CiscoSystemsCCM-SIP 173064 1 IN IP4 172.16.200.20
s=SIP Call
c=IN IP4 172.16.3.158
b=TIAS:64000
b=AS:64
t=0 0
m=audio 19490 RTP/AVP 8 9 18 0 116 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:116 iLBC/8000
a=ptime:20
a=maxptime:60
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Outgoing dial-peer configuration:
=========================
dial-peer voice 802 voip
description ** SIP TO STC **
translation-profile outgoing OUT
destination-pattern .T
rtp payload-type cisco-codec-fax-ack 98
rtp payload-type nte 97
session protocol sipv2
session target ipv4:10.200.7.157:5060
session transport udp
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
Incoming translation rule/dial-peer
==========================
dial-peer voice 202 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIPINCOMING283
translation-profile outgoing REDIAL283
preference 1
destination-pattern 5...
session protocol sipv2
session target ipv4:172.16.200.13
incoming called-number 2835...$
voice-class codec 1
dtmf-relay sip-notify rtp-nte sip-kpml
no vad
!
dial-peer voice 301 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIPINCOMING243
translation-profile outgoing REDIAL243
destination-pattern 8...
session protocol sipv2
session target ipv4:172.16.200.20
incoming called-number 2438...$
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay sip-notify rtp-nte sip-kpml
no vad
voice translation-profile SIPINCOMING243
translate called 152
!
voice translation-rule 152
rule 1 /^2438/ /8/
rule 2 /^12438/ /8/
rule 3 /^112438/ /8/
rule 4 /^0112438/ /8/
voice translation-profile SIPINCOMING283
translate called 52
voice translation-rule 52
rule 1 /^2835/ /5/
rule 2 /^12835/ /5/
rule 3 /^112835/ /5/
rule 4 /^0112835/ /5/
I also see a significant change between configuration of 2835XXX/ 2438XXX and 4945XXX. In case of later, you don't have a customized incoming dial-peer that translation rule reducing seven digit number to 4 digits. Is there a change in calling number requirement from telco?
Thanks!
Kunal
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