[Asterisk-Users] SIP Debug

[Michael Anuzis]

Using Asterisk Management Portal with Broadvoice. It used to work just
fine; calls would come in and be answered with no trouble at all. A
few weeks ago with no configuration changes at all Asterisk stopped
picking up calls and started giving a busy signal whenever someone
calls.    I've tried rebooting the system many times, and "sip show
registry" shows it's registering correctly with Broadvoice. Sip debug
shows the UDP packets correctly hit the system on port 5060, but the
call is rejected\busy instead of answered.

Here's a SIP debug or a call coming in and being busy. Any clues? 

Sip read:
INVITE sip:XXXX at 192.168.1.107:5060 SIP/2.0
Call-ID: ff01aa-43 at 147.135.12.128
CSeq: 1 INVITE
From: "XXXX"<sip:XXXX at 147.135.12.128;user=phone>;tag=xz13
To: "XXXX"<sip:s at 192.168.1.107;user=phone>
Via: SIP/2.0/UDP 147.135.12.128:5060
Contact: sip:XXXX at 147.135.12.128:5060
Supported: 100rel
RPID-Privacy: party=calling;id-type=subscriber;privacy=off
Remote-Party-ID: <sip:XXXX 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 18092 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 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 147.135.12.250:18092
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)
Found peer 'sip.broadvoice.com'
Looking for XXXX in from-pstn
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 147.135.12.128:5060
From: "XXXX"<sip:XXXX at 147.135.12.128;user=phone>;tag=xz13
To: "XXXX"<sip:s at 192.168.1.107;user=phone>;tag=as54c1e248
Call-ID: ff01aa-43 at 147.135.12.128
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:XXXX at 192.168.1.107>
Content-Length: 0


 to 147.135.12.128:5060
asterisk1*CLI>