[Asterisk-Users] SIP Debug

[Michael Anuzis]

Using Asterisk Management Portal, this config used to work just fine,
but it randomly stopped working a few weeks ago. sip show registry
shows the number is registering correctly with Broadvoice, & sip debug
shows calls coming in but they always get a busy signal. Any idea
what's going on? Here's a sip debug:

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>