I’m not exactly sure what the problem here is. I’m getting a 404 on the DID Number it seems. I have my inbound route DID Number set to my SIP-ID, and set to an extension as my destination. I can place outbound calls just fine.
<— SIP read from 204.155.28.10:5060 —>
INVITE sip:MYSIPID@208.125.123.163 SIP/2.0
Record-Route:
Record-Route:
Record-Route:
Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bK7fdf.a2434d06.0
Via: SIP/2.0/UDP 172.30.20.1;branch=z9hG4bK7fdf.a2434d06.0
Via: SIP/2.0/UDP 204.155.28.10:5060;received=204.155.28.20;branch=z9hG4bK5dfb9758
Via: SIP/2.0/UDP 204.155.29.58:5060;branch=z9hG4bK5dfb9758;rport=5060
Max-Forwards: 67
From: “4156398597″
To:
Contact:
Call-ID: 319761ae61e4ce6c7bc6d4512dfa3da1@sipgate.com
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-Timeout: 120
Content-Type: application/sdp
Content-Length: 383
v=0
o=root 336087564 336087565 IN IP4 204.155.29.58
s=sipgate VoIP GW
c=IN IP4 204.155.29.58
t=0 0
m=audio 19352 RTP/AVP 0 8 3 18 112 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<————->
— (19 headers 17 lines) —
Sending to 204.155.28.10 : 5060 (NAT)
Using INVITE request as basis request - 319761ae61e4ce6c7bc6d4512dfa3da1@sipgate.com
Found peer ‘SipGate’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 112
Found RTP audio format 101
Peer audio RTP is at port 204.155.29.58:19352
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 112
Found audio description format telephone-event for ID 101
Capabilities: us - 0×8 (alaw), peer - audio=0×90e (gsm|ulaw|alaw|g726|g729)/video=0×0 (nothing), combined - 0×8 (alaw)
Non-codec capabilities (dtmf): us - 0×1 (telephone-event), peer - 0×1 (telephone-event), combined - 0×1 (telephone-event)
Peer audio RTP is at port 204.155.29.58:19352
Looking for MYSIPID in ext_did (domain 208.125.123.163)
trixbox*CLI>
<— Reliably Transmitting (NAT) to 204.155.28.10:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bK7fdf.a2434d06.0;received=204.155.28.10
Via: SIP/2.0/UDP 172.30.20.1;branch=z9hG4bK7fdf.a2434d06.0
Via: SIP/2.0/UDP 204.155.28.10:5060;received=204.155.28.20;branch=z9hG4bK5dfb9758
Via: SIP/2.0/UDP 204.155.29.58:5060;branch=z9hG4bK5dfb9758;rport=5060
From: “4156398597″
To:
Call-ID: 319761ae61e4ce6c7bc6d4512dfa3da1@sipgate.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<————>
Scheduling destruction of SIP dialog ‘319761ae61e4ce6c7bc6d4512dfa3da1@sipgate.com’ in 7424 ms (Method: INVITE)
trixbox*CLI>
<— SIP read from 204.155.28.10:5060 —>
ACK sip:MYSIPID@208.125.123.163 SIP/2.0
Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bK7fdf.a2434d06.0
Via: SIP/2.0/UDP 172.30.20.1;branch=z9hG4bK7fdf.a2434d06.0
From: “4156398597″
Call-ID: 319761ae61e4ce6c7bc6d4512dfa3da1@sipgate.com
To:
CSeq: 103 ACK
Max-Forwards: 69
Content-Length: 0
X-hint: rr-enforced
<————->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘319761ae61e4ce6c7bc6d4512dfa3da1@sipgate.com’ Method: ACK
Really destroying SIP dialog ‘9e3ab88cf8ba88ce’ Method: REGISTER
Really destroying SIP dialog ‘e1e50191017e798e’ Method: REGISTER
