Hi!
I have CUBE between ITSP and CUCM. Incoming call from ITSP yave field TO with alphanumeric username, like this:
INVITE sip:VSN8101148001@192.168.111.254:5060 SIP/2.0
Max-Forwards: 68
Via: SIP/2.0/UDP 195.239.174.100:5060;branch=z9hG4bKg3Zqkv7itmtc4u8x25kreq6omsr9sgj68
To: "SIP-201 SIP-201" <sip:VSN8101148001@sip.beeline.ru>
From: <sip:74999298101@sip.beeline.ru;user=phone>;tag=h7g4Esbg_283817668-1425470133849-
Call-ID: BW1455338490403151587754552@10.64.248.6
CSeq: 844237613 INVITE
Contact: <sip:sgc_c@195.239.174.100;transport=udp>
Record-Route: <sip:195.239.174.100;transport=udp;lr>
Min-Se: 180
P-Asserted-Identity: <sip:+74999298101@sip.beeline.ru;user=phone>
Privacy: none
Session-Expires: 1800
Supported: 100rel
Supported: timer
Content-Type: application/sdp
Content-Length: 260
Recv-Info: x-broadworks-client-session-info
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-hotsip-filetransfer+xml
Accept: multipart/mixed
v=0
o=BroadWorks 109455546 1 IN IP4 195.239.174.100
s=-
c=IN IP4 195.239.174.71
t=0 0
m=audio 41410 RTP/AVP 8 0 18 96 101
a=fmtp:18 annexb=no
a=rtpmap:96 AMR/8000
a=fmtp:96 octet-align=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
And this username is big problem! How i can translate this invite to CUCM with DN number? I use:
!
voice class sip-profiles 3
request INVITE sip-header To modify "sip:VSN8101148001" "sip:4957973503"
!
!
voice class uri BEELINE sip
host dns:sip.beeline.ru
dial-peer voice 100 voip
description From_Beeline_to_CUCM
session protocol sipv2
incoming uri from BEELINE
voice-class codec 10
voice-class sip call-route url
voice-class sip profiles 3
!
!
dial-peer voice 102 voip
destination-pattern 4957973503
session protocol sipv2
session target ipv4:192.168.253.2
voice-class codec 10
voice-class sip early-offer forced
dtmf-relay rtp-nte sip-notify sip-kpml h245-alphanumeric h245-signal cisco-rtp
!
And this is not work. CUBE terminate incoming call to 100 dial-peer, and can't find outgoing dial-peer.
