[Asterisk-Users] broadvoice config problem.

[Rich Adamson]

> 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=SD5oc1901-171810273
0-1113256175088
> 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

Looks to me like Broadvoice is trying to authenticate a call with
your system and can only find g723 codec (which doesn't exist in *). 
Then it looks like its trying to dial x100. Do you have a context 
called "from-broadvoice" that includes something like:
 exten => 100,1,Dial(SIP/3001,15,r) 

So, not sure whether your result is an incorrect codec, an exten=>100
problem, or both.



> 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=SD5oc1901-171810273
0-1113256175088
> 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=SD5oc1901-171810273
0-1113256175088
> 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
> 
> 
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