Hey Guys,
Been a while since ive done one of these. I have an issue where CUE isnt picking up the called number correctly, but only for SIP calls.
If I call into an SCCP Phone on extension 218, the call comes in ok and gets VM.
If I call into a SIP phone on extension 210, the call goes to VM Ok, but asks for the ID.
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/0.3
bind media source-interface GigabitEthernet0/0.3
registrar server expires max 600 min 60
voice register global
mode cme
source-address x.x.x.x port 5060
max-dn 20
max-pool 20
load 8961 sip8961.9-4-1-9
timezone 23
time-format 24
date-format D/M/Y
voicemail 700
tftp-path flash:
create profile sync 0006489029377307
!
voice register dn 1
number 210
allow watch
name Reception
call-forward b2bua all 700
!
voice register template 1
camera
video
!
voice register pool 1
busy-trigger-per-button 2
id mac 189C.5D21.D727
type 8961 addon 1 CKEM
number 1 dn 1
template 1
dtmf-relay rtp-nte sip-notify
username cisco password cisco
dial-peer voice 700 voip
mailbox-selection orig-called-num
description *** VM ***
destination-pattern 700
session protocol sipv2
session target ipv4:10.107.41.201
dtmf-relay rtp-nte sip-notify
codec g711ulaw
no vad
Doing a debug CCSIP messsages shows that with the SCCP call debug, there is a "Diversion" item in the SIP invite to the CUE:
INVITE sip:700@10.107.41.201:5060 SIP/2.0
Via: SIP/2.0/UDP 10.107.41.200:5060;branch=z9hG4bK1F27D1D1F
Remote-Party-ID: <sip:000447590695490@10.107.41.200>;party=calling;screen=yes;privacy=off
From: <sip:000447590695490@10.107.41.200>;tag=D5D50D48-1678
To: <sip:700@10.107.41.201>
Date: Tue, 14 Jul 2015 15:31:20 GMT
Call-ID: 35DEA64F-297411E5-80B8BE43-45F64916@10.107.41.200
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0903744039-0695472613-2159328835-1173768470
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1436887880
Contact: <sip:000447590695490@10.107.41.200:5060>
Call-Info: <sip:10.107.41.200:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Diversion: <sip:218@10.107.41.200>;privacy=off;reason=unconditional;counter=1;screen=no
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 250
v=0
o=CiscoSystemsSIP-GW-UserAgent 4399 3556 IN IP4 10.107.41.200
s=SIP Call
c=IN IP4 10.107.41.200
t=0 0
m=audio 20248 RTP/AVP 0 101
c=IN IP4 10.107.41.200
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
However, the call trace when involving a SIP based first hop telephone does not show this:
INVITE sip:700@10.107.41.201:5060 SIP/2.0
Via: SIP/2.0/UDP 10.107.41.200:5060;branch=z9hG4bK1F2831903
Remote-Party-ID: <sip:000447590695490@10.107.41.200>;party=calling;screen=yes;privacy=off
From: <sip:000447590695490@10.107.41.200>;tag=D5D6B854-1A4C
To: <sip:700@10.107.41.201>
Date: Tue, 14 Jul 2015 15:33:09 GMT
Call-ID: 771064ED-297411E5-80C6BE43-45F64916@10.107.41.200
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1997523141-0695472613-2160311875-1173768470
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1436887989
Contact: <sip:000447590695490@10.107.41.200:5060>
Call-Info: <sip:10.107.41.200:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 250
v=0
o=CiscoSystemsSIP-GW-UserAgent 6051 2043 IN IP4 10.107.41.200
s=SIP Call
c=IN IP4 10.107.41.200
t=0 0
m=audio 20250 RTP/AVP 0 101
c=IN IP4 10.107.41.200
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
I feel like I need the equivalent of "redirecting diversion header delivery - inbound" for CCME - does anyone know how to solve this?
Thanks
