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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 147.135.8.128:5060;branch=z9hG4bK2imkag00d031s9k3e1k1.1sr
From: "Simon Craig"<sip:[EMAIL PROTECTED];user=phone;bvoice=ACME-06t5tpji5ub7e>;tag=SD5oc1901-1718102730-1113256175088
To: "Craig Simon"<sip:[EMAIL PROTECTED];user=phone>
Call-ID: SD5oc1901-e89dc59f2c879f7144697a9486593537-js1h002
CSeq: 429822713 INVITE
Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED];user=phone;bvoice=ACME-06t5tpji5ub7e>;tag=SD5oc1901-1718102730-1113256175088
To: "Craig Simon"<sip:[EMAIL PROTECTED];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:[EMAIL PROTECTED]>
Content-Length: 0
to 147.135.8.128:5060 asterisk*CLI>
Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 147.135.8.128:5060;branch=z9hG4bK2imkag00d031s9k3e1k1.1sr
From: "Simon Craig"<sip:[EMAIL PROTECTED];user=phone;bvoice=ACME-06t5tpji5ub7e>;tag=SD5oc1901-1718102730-1113256175088
To: "Craig Simon"<sip:[EMAIL PROTECTED];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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