Hello. Here is the invite. On this number there is just the voicemail.
Code:
INVITE sip:190475@64.34.163.35:5061 SIP/2.0
Via: SIP/2.0/UDP 212.109.160.59:5060;branch=z9hG4bK2d21db3b;rport
Route:
<sip:64.34.173.199;lr=on;ftag=as75fcf5be>,<sip:64.34.163.35;lr=on;ftag=as75fcf5be>
From: "1001" <sip:1001@omnianet.it>;tag=as75fcf5be
To: <sip:190475@voxalot.com>;tag=as2d2dbc73
Contact: <sip:1001@212.109.160.59>
Call-ID: 5478b0226d9238c54aaf9ea052283918@omnianet.it
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 309
v=0
o=root 2200 2201 IN IP4 192.168.10.65
s=session
c=IN IP4 192.168.10.65
t=0 0
m=audio 16152 RTP/AVP 0 8 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv