[Asterisk-Users] broadvoice config problem.

[Craig Simon]

Hello list

I have been fighting with * for a couple of days now.  I have recieved 
some help from the list but have not been successful in receiving calls 
from broadvoice to my asterisk box yet.  I can place calls however, just 
not receive them.  I enabled sip debugging today and here is the output 
from an incoming call:

asterisk*CLI> sip debug
SIP Debugging Enabled
asterisk*CLI>

Sip read:
INVITE sip:100 at 207.145.49.194 SIP/2.0
Via: SIP/2.0/UDP 147.135.8.128:5060;branch=z9hG4bK2imkag00d031s9k3e1k1.1sr
From: "Simon 
Craig"<sip:9251234567 at 147.135.8.129;user=phone;bvoice=ACME-06t5tpji5ub7e>;tag=SD5oc1901-1718102730-1113256175088
To: "Craig Simon"<sip:9255582025 at sip.broadvoice.com;user=phone>
Call-ID: SD5oc1901-e89dc59f2c879f7144697a9486593537-js1h002
CSeq: 429822713 INVITE
Contact: 
<sip:9251234567 at 147.135.8.128:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY
Supported: 100rel,timer
Min-SE: 60
Accept: application/sdp,application/dtmf
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 289

v=0
o=BroadWorks 3802511 1 IN IP4 147.135.8.128
s=-
c=IN IP4 147.135.8.128
t=0 0
m=audio 14022 RTP/AVP 0 8 96 18 97 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000

14 headers, 13 lines
Using latest request as basis request
Sending to 147.135.8.128 : 5060 (NAT)
Found peer 'sip.broadvoice.com'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 18
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 147.135.8.128:14022
Found description format PCMU
Found description format PCMA
Found description format G726-32
Found description format G729
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x51c 
(ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 
0x1 (g723)
Looking for 100 in from-broadvoice
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 147.135.8.128:5060;branch=z9hG4bK2imkag00d031s9k3e1k1.1sr
From: "Simon 
Craig"<sip:9251234567 at 147.135.8.129;user=phone;bvoice=ACME-06t5tpji5ub7e>;tag=SD5oc1901-1718102730-1113256175088
To: "Craig 
Simon"<sip:9255582025 at sip.broadvoice.com;user=phone>;tag=as1fe9ff98
Call-ID: SD5oc1901-e89dc59f2c879f7144697a9486593537-js1h002
CSeq: 429822713 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:100 at 207.145.49.194>
Content-Length: 0


 to 147.135.8.128:5060
asterisk*CLI>

Sip read:
ACK sip:100 at 207.145.49.194 SIP/2.0
Via: SIP/2.0/UDP 147.135.8.128:5060;branch=z9hG4bK2imkag00d031s9k3e1k1.1sr
From: "Simon 
Craig"<sip:9251234567 at 147.135.8.129;user=phone;bvoice=ACME-06t5tpji5ub7e>;tag=SD5oc1901-1718102730-1113256175088
To: "Craig 
Simon"<sip:9255582025 at sip.broadvoice.com;user=phone>;tag=as1fe9ff98
Call-ID: SD5oc1901-e89dc59f2c879f7144697a9486593537-js1h002
CSeq: 429822713 ACK


6 headers, 0 lines
Destroying call 'SD5oc1901-e89dc59f2c879f7144697a9486593537-js1h002'
asterisk*CLI>


I see the 404 in the middle of the log, I just am not sure what it is 
looking for and not finding.  Any help would be great.

Thanks
Craig