En ruta entrante tengo +17864716984. Y el Outbound CallerID que me da el proveedor es +17864716984. En el debug veo que la llamada llega a mi central:
<--- SIP read from UDP:209.208.211.36:5060 --->
INVITE sip:+17864716984@10.0.0.20:5060 SIP/2.0
Via: SIP/2.0/UDP 209.208.211.36:5060;branch=z9hG4bK2cf63b476cf850
From: "unknown" <sip:+17863422059@209.208.211.36>;tag=9883768~7ac60a5e-4e44-4649-b651-52cc71805e7a-43698241
To: <sip:+17864716984@10.0.0.20>
Date: Thu, 08 Jun 2017 17:06:13 GMT
Call-ID: c55e3600-93918405-16ba0d-24d3d0d1@209.208.211.36
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM11.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=MIXED
Session-ID: 40bec392be0f3302881f4fcab9883767;remote=00000000000000000000000000000000
Cisco-Guid: 3311285760-0000065536-0000085035-0617861329
Session-Expires: 86400
P-Asserted-Identity: "unknown" <sip:+17863422059@209.208.211.36>
Remote-Party-ID: "unknown" <sip:+17863422059@209.208.211.36>;party=calling;screen=yes;privacy=off
Contact: <sip:+17863422059@209.208.211.36:5060>
Max-Forwards: 67
Content-Type: application/sdp
Content-Length: 288
v=0
o=CiscoSystemsCCM-SIP 9883768 1 IN IP4 209.208.211.36
s=SIP Call
c=IN IP4 38.105.10.195
b=TIAS:64000
b=AS:64
t=0 0
m=audio 33594 RTP/AVP 0 18 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (24 headers 14 lines) ---
Sending to 209.208.211.36:5060 (NAT)
Sending to 209.208.211.36:5060 (NAT)
Using INVITE request as basis request - c55e3600-93918405-16ba0d-24d3d0d1@209.208.211.36
Found peer 'Mexico' for '+17863422059' from 209.208.211.36:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 38.105.10.195:33594
Looking for +17864716984 in from-internal (domain 10.0.0.20)
<--- Reliably Transmitting (no NAT) to 209.208.211.36:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 209.208.211.36:5060;branch=z9hG4bK2cf63b476cf850;received=209.208.211.36
From: "unknown" <sip:+17863422059@209.208.211.36>;tag=9883768~7ac60a5e-4e44-4649-b651-52cc71805e7a-43698241
To: <sip:+17864716984@10.0.0.20>;tag=as59a873e4
Call-ID: c55e3600-93918405-16ba0d-24d3d0d1@209.208.211.36
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
[% F% T] NOTICE[14306][C-000121fd]: chan_sip.c:25872 handle_request_invite: Call from 'Mexico' (209.208.211.36:5060) to extension '+17864716984' rejected because extension not found in context 'from-internal'.
Scheduling destruction of SIP dialog 'c55e3600-93918405-16ba0d-24d3d0d1@209.208.211.36' in 6400 ms (Method: INVITE)
Retransmitting #1 (no NAT) to 209.208.211.36:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 209.208.211.36:5060;branch=z9hG4bK2cf63b476cf850;received=209.208.211.36
From: "unknown" <sip:+17863422059@209.208.211.36>;tag=9883768~7ac60a5e-4e44-4649-b651-52cc71805e7a-43698241
To: <sip:+17864716984@10.0.0.20>;tag=as59a873e4
Call-ID: c55e3600-93918405-16ba0d-24d3d0d1@209.208.211.36
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0