[Asterisk-Users] Need Help Troubleshooting Broadvoice Connection

[Tim P]

Ok it seems that the pbx can see that I am recieving a call (or at
least my broadvoice number sees it I'm not sure which)

Here is  the results of me making a call to my pbx with "sip debug
peer bv" (broadvoice)
Can someone please take a look at this output, it looks like the call
is recieved but either not acted upon or something.  All calls get a
fast buys and broadvoice claims it isn't them.  I have all firewall
ports open that need to be (5060-5070 udp+tcp, 10000-20000 udp, 69
udp)
The call originates from me (Tim Porritt) to the number registered
with Broadvoice (Kira Duckett), any idea of the issue?  Everything
looks fine as far as I can see.

Sip read:
INVITE sip:2068660133 at 192.168.8.151:5060 SIP/2.0
Call-ID: ff0376-37 at 147.135.12.128
CSeq: 1 INVITE
From: "Porritt Tim"<sip:9417277118 at 147.135.12.128;user=phone>;tag=9bdf
To: "Kira Duckett"<sip:s at 192.168.8.151;user=phone>
Via: SIP/2.0/UDP 147.135.12.128:5060
Contact: sip:9417277118 at 147.135.12.128:5060
Supported: 100rel
RPID-Privacy: party=calling;id-type=subscriber;privacy=off
Remote-Party-ID:
<sip:9417277118 at 147.135.12.128>;screen=yes;party=calling;privacy=off
Content-Length:  273
Content-Type: application/sdp

v=0
o=2475101431 10 10 IN IP4 147.135.12.247
s=-
c=IN IP4 147.135.12.250
t=0 0
m=audio 33532 RTP/AVP 0 8 2 18 96 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 iLBC/8000
a=rtpmap:101 telephone-event/8000

12 headers, 12 lines
Using latest request as basis request
Sending to 147.135.12.128 : 5060 (non-NAT)
Found peer 'bv'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 101
Peer audio RTP is at port 147.135.12.250:33532
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 - 0xc (ulaw|alaw), 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 2068660133 in from-pstn
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 147.135.12.128:5060
From: "Porritt Tim"<sip:9417277118 at 147.135.12.128;user=phone>;tag=9bdf
To: "Kira Duckett"<sip:s at 192.168.8.151;user=phone>;tag=as6fafa40c
Call-ID: ff0376-37 at 147.135.12.128
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2068660133 at 192.168.8.151>
Content-Length: 0


 to 147.135.12.128:5060
asterisk1*CLI>

Sip read:
ACK sip:s at 192.168.8.151:5060 SIP/2.0
Call-ID: ff0376-37 at 147.135.12.128
CSeq: 1 ACK
From: "Porritt Tim"<sip:9417277118 at 147.135.12.128;user=phone>;tag=9bdf
To: "Kira Duckett"<sip:s at 192.168.8.151;user=phone>;tag=as6fafa40c
Via: SIP/2.0/UDP 147.135.12.128:5060;received=24.17.77.152
Content-Length:    0


7 headers, 0 lines
Destroying call 'ff0376-37 at 147.135.12.128'
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager_custom.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.8.151:5060;branch=z9hG4bK4e5a6712
From: <sip:2068660133 at sip.broadvoice.com>;tag=as3111bfd4
To: <sip:2068660133 at sip.broadvoice.com>
Call-ID: 37c0481f10a3d18356f5dcfc0021fe7b at 127.0.0.1
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:s at 192.168.8.151>
Event: registration
Content-Length: 0