[Asterisk-Users] Can someone tell me what is going on from this debug?

[Doug Eubanks]

Can someone tell me why Asterisk is sending 404 instead of passing this call to the demo?  I have replaced the IPs with descriptions....

This is the actual asterisk debug,

Thanks
Doug Eubanks
doug at simflex.com

Sip read:
INVITE sip:19995551212 at ASTERISKSERVER:5060 SIP/2.0
Via: SIP/2.0/UDP GATEKEEPER:5060;branch=z9hG4bK-506011002837883257484230
Via: SIP/2.0/UDP ANALOGADAPTER:5060;branch=z9hG4bK-9a1dbb20
From: "19993152553" <sip:19993152553 at GATEKEEPER>;tag=9d074f5b40d37cf
To: <sip:19995551212 at GATEKEEPER>
Call-ID: 60b8452f-d6a70d48 at ANALOGADAPTER
CSeq: 101 INVITE
Max-Forwards: 69
Contact: "19993152553" <sip:19993152553 at GATEKEEPER:5060>
Expires: 240
User-agent: Sipura/SPA2000-1.0.37(e)
Content-Length: 399
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
Record-Route: <sip:GATEKEEPER:5060;lr>

v=0
o=- 292396 292396 IN IP4 GATEKEEPER
s=-
c=IN IP4 GATEKEEPER
t=0 0
m=audio 1690 RTP/AVP 0 8 96 2 97 98 18 101 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:18 G729a/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 NSE/8000
a=ptime:30
a=sendrecv

15 headers, 18 lines
Using latest request as basis request
Sending to GATEKEEPER : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 18
Found RTP audio format 101
Found RTP audio format 100
Peer audio RTP is at port GATEKEEPER:1690
Found description format PCMU
Found description format PCMA
Found description format G726-40
Found description format G726-32
Found description format G726-24
Found description format G726-16
Found description format G729a
Found description format telephone-event
Found description format NSE
Capabilities: us - 0x60e(GSM|ULAW|ALAW|SPEEX|ILBC), peer - audio=0x51c(ULAW|ALAW|G726|G729A|ILBC)/video=0x0(EMPTY), combined - 0x40c(ULAW|ALAW|ILBC)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
Found peer 'sip.simflex.net'
Looking for 19995551212 in default
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP GATEKEEPER:5060;branch=z9hG4bK-506011002837883257484230
Via: SIP/2.0/UDP ANALOGADAPTER:5060;branch=z9hG4bK-9a1dbb20
From: "19993152553" <sip:19993152553 at GATEKEEPER>;tag=9d074f5b40d37cf
To: <sip:19995551212 at GATEKEEPER>;tag=as2a8cca97
Call-ID: 60b8452f-d6a70d48 at ANALOGADAPTER
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:19995551212 at ASTERISKSERVER>
Content-Length: 0


 to GATEKEEPER:5060
DomainMailV3*CLI>

Sip read:
ACK sip:19995551212 at ASTERISKSERVER:5060 SIP/2.0
Via: SIP/2.0/UDP GATEKEEPER:5060;branch=z9hG4bK-506011002837893257484230
Via: SIP/2.0/UDP ANALOGADAPTER:5060;branch=z9hG4bK-9a1dbb20
From: "19993152553" <sip:19993152553 at GATEKEEPER>;tag=9d074f5b40d37cf
To: <sip:19995551212 at GATEKEEPER>;tag=as2a8cca97
Call-ID: 60b8452f-d6a70d48 at ANALOGADAPTER
CSeq: 101 ACK
Max-Forwards: 69
Contact: "19993152553" <sip:19993152553 at GATEKEEPER:5060>
User-agent: Sipura/SPA2000-1.0.37(e)
Content-Length: 0


11 headers, 0 lines
Destroying call '60b8452f-d6a70d48 at ANALOGADAPTER'



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