[Asterisk-Dev] Bug in realtime SIP

Fri Aug 12 20:49:26 MST 2005

Asterisk version:  CVS Head from 8/11/05

extconfig.conf

sipusers => mysql,zbx,sipusers
sippeers => mysql,zbx,sippeers

localhost*CLI> realtime mysql status
Connected to zbx at localhost, port 3306 with username zbx for 27 minutes,
20 seconds.

realtime load sippeers name 104 and realtime load sipusers 104 both show
the sip device information on screen.

When the sip device registers (it's dynamic), sip show peers and sip
show users show the following

sip show peers
104/104                    (Unspecified)    D          255.255.255.255 
0        Unmonitored

sip show users

No entry every shows up for users.  If I try to make a call with the
device, I get 403 Forbidden

<-- SIP read from 192.168.254.16:5060:
INVITE sip:18005551212 at 192.168.254.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.16:5060;branch=z9hG4bK-ba65de3c
From: Sipura2000-2 <sip:104 at 192.168.254.6>;tag=ffbcad7047bef41co1
To: <sip:18005551212 at 192.168.254.6>
Call-ID: 11739700-af5297cc at 192.168.254.16
CSeq: 101 INVITE
Max-Forwards: 70
Contact: Sipura2000-2 <sip:104 at 192.168.254.16:5060>
Expires: 240
User-Agent: Sipura/SPA2000-2.0.10(c)
Content-Length: 426
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 266317 266317 IN IP4 192.168.254.16
s=-
c=IN IP4 192.168.254.16
t=0 0
m=audio 16384 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (14 headers 19 lines)---
Using INVITE request as basis request - 11739700-af5297cc at 192.168.254.16
Sending to 192.168.254.16 : 5060 (non-NAT)
Found no matching peer or user for '192.168.254.16:5060'
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.254.16:16384
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d
(g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404
(ulaw|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 18005551212 in default
list_route: hop: <sip:104 at 192.168.254.16:5060>
Transmitting (no NAT) to 192.168.254.16:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.254.16:5060;branch=z9hG4bK-ba65de3c
From: Sipura2000-2 <sip:104 at 192.168.254.6>;tag=ffbcad7047bef41co1
To: <sip:18005551212 at 192.168.254.6>
Call-ID: 11739700-af5297cc at 192.168.254.16
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:18005551212 at 192.168.254.4>
Content-Length: 0

---
    -- Executing Hangup("SIP/192.168.254.6-08a17090", "") in new stack
  == Spawn extension (default, 18005551212, 1) exited non-zero on
'SIP/192.168.254.6-08a17090'
Reliably Transmitting (no NAT) to 192.168.254.16:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.254.16:5060;branch=z9hG4bK-ba65de3c
From: Sipura2000-2 <sip:104 at 192.168.254.6>;tag=ffbcad7047bef41co1
To: <sip:18005551212 at 192.168.254.6>;tag=as31d7c52b
Call-ID: 11739700-af5297cc at 192.168.254.16
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:18005551212 at 192.168.254.4>
Content-Length: 0