[Asterisk-Users] Need Help Troubleshooting Broadvoice Connection

[Tim P]

Problem solved:
Needed to append a /2068660133  to the end of the register string (my
phone number) and the create a did route with that number.  At that
point it worked fine.

On 8/8/05, Tim P <panterafreak at gmail.com> wrote:
> 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
>