> 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@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@147.135.8.129;user=phone;bvoice=ACME-06t5tpji5ub7e>;tag=SD5oc1901-171810273
0-1113256175088> To: "Craig
Simon"<sip:9255582025@sip.broadvoice.com;user=phone>
> Call-ID: SD5oc1901-e89dc59f2c879f7144697a9486593537-js1h002
> CSeq: 429822713 INVITE
> Contact:
>
<sip:9251234567@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@147.135.8.129;user=phone;bvoice=ACME-06t5tpji5ub7e>;tag=SD5oc1901-171810273
0-1113256175088> To: "Craig
>
Simon"<sip:9255582025@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@207.145.49.194>
> Content-Length: 0
>
>
> to 147.135.8.128:5060
> asterisk*CLI>
>
> Sip read:
> ACK sip:100@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@147.135.8.129;user=phone;bvoice=ACME-06t5tpji5ub7e>;tag=SD5oc1901-171810273
0-1113256175088> To: "Craig
>
Simon"<sip:9255582025@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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