[Asterisk-Users] sip show registry empty ?!?!!?

Michele "O-Zone" Pinassi zerozone.liste at gmail.com
Tue May 17 03:58:34 MST 2005 

Hi all,
i've installed Asterisk with AMP. I've created 4 extensions (for 4 SIP phones) 
and this is what my "sip show users" return:

moloch*CLI> sip show users
Username         Secret           Accountcode     Def.Context     ACL  NAT
204              moira                            from-internal   No   No
203              michele                          from-internal   No   No
202              duccio                           from-internal   No   No
201              fabrizio                         from-internal   No   No
moloch*CLI>                    

it's ok. So i use kphone to connect top my asterisk server. KPhone say that 
i'm on-line so i'll check "sip show registry" and it's empty:

moloch*CLI> sip show registry
Host                            Username       Refresh State
moloch*CLI>    

If i try, from 203, calling 201 this is what happens:

===========================8<===================================

moloch*CLI>

Sip read:
INVITE sip:201 at asb.unisi.it SIP/2.0
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK78ED0E83
CSeq: 7665 INVITE
To: <sip:201 at asb.unisi.it>
Content-Type: application/sdp
From: "203" <sip:203 at asb.unisi.it>;tag=2B558754
Call-ID: 499575437 at 192.167.125.9
Subject: sip:203 at asb.unisi.it
Content-Length: 187
User-Agent: kphone/4.0.5
Contact: "203" <sip:203 at 192.167.125.9:5062;transport=udp>

v=0
o=username 0 0 IN IP4 192.167.125.9
s=The Funky Flow
c=IN IP4 192.167.125.9
t=0 0
m=audio 35996 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000

11 headers, 9 lines
Using latest request as basis request
Sending to 192.167.125.9 : 5062 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK78ED0E83
From: "203" <sip:203 at asb.unisi.it>;tag=2B558754
To: <sip:201 at asb.unisi.it>;tag=as17f37979
Call-ID: 499575437 at 192.167.125.9
CSeq: 7665 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:201 at 192.167.125.9>
Proxy-Authenticate: Digest realm="asterisk", nonce="2149fad7"
Content-Length: 0


 to 192.167.125.9:5062
Scheduling destruction of call '499575437 at 192.167.125.9' in 15000 ms
Found user '203'
moloch*CLI>

Sip read:
ACK sip:201 at asb.unisi.it SIP/2.0
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK78ED0E83
CSeq: 7665 ACK
To: <sip:201 at asb.unisi.it>;tag=as17f37979
From: "203" <sip:203 at asb.unisi.it>;tag=2B558754
Call-ID: 499575437 at 192.167.125.9
Content-Length: 0
User-Agent: kphone/4.0.5
Contact: "203" <sip:203 at 192.167.125.9:5062;transport=udp>


9 headers, 0 lines
moloch*CLI>

Sip read:
INVITE sip:201 at asb.unisi.it SIP/2.0
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK4236106
CSeq: 7666 INVITE
To: <sip:201 at asb.unisi.it>
Proxy-Authorization: Digest username="203", realm="asterisk", 
nonce="2149fad7", uri="sip:201 at asb.unisi.it", cnonce="abcdefghi", 
nc=00000001, response="b1a9c4ee2ac7065635f681a281dcec25", opaque="", 
algorithm="MD5"
Content-Type: application/sdp
From: "203" <sip:203 at asb.unisi.it>;tag=2B558754
Call-ID: 499575437 at 192.167.125.9
Subject: sip:203 at asb.unisi.it
Content-Length: 187
User-Agent: kphone/4.0.5
Contact: "203" <sip:203 at 192.167.125.9:5062;transport=udp>

v=0
o=username 0 0 IN IP4 192.167.125.9
s=The Funky Flow
c=IN IP4 192.167.125.9
t=0 0
m=audio 35996 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000

12 headers, 9 lines
Using latest request as basis request
Sending to 192.167.125.9 : 5062 (non-NAT)
Found user '203'
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 3
Peer audio RTP is at port 192.167.125.9:35996
Found description format PCMU
Found description format GSM
Found description format iLBC
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x406 (gsm|ulaw|
ilbc)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 
(nothing)
Looking for 201 in from-internal
list_route: hop: <sip:203 at 192.167.125.9:5062;transport=udp>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK4236106
From: "203" <sip:203 at asb.unisi.it>;tag=2B558754
To: <sip:201 at asb.unisi.it>
Call-ID: 499575437 at 192.167.125.9
CSeq: 7666 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:201 at 192.167.125.9>
Content-Length: 0


 to 192.167.125.9:5062
    -- Executing Macro("SIP/203-14e6", "exten-vm|201 at default|201") in new 
stack
    -- Executing SetVar("SIP/203-14e6", "FROMCONTEXT=exten-vm") in new stack
    -- Executing GotoIf("SIP/203-14e6", "0?novm|1:3") in new stack
    -- Goto (macro-exten-vm,s,3)
    -- Executing GotoIf("SIP/203-14e6", "0?novm|1") in new stack
    -- Executing Macro("SIP/203-14e6", "dial|15|tr|201") in new stack
    -- Executing AGI("SIP/203-14e6", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing Wait("SIP/203-14e6", "1") in new stack
    -- Executing VoiceMail("SIP/203-14e6", "u201 at default") in new stack
We're at 192.167.125.9 port 18376
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with preferred capability 0x2 (gsm)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK4236106
From: "203" <sip:203 at asb.unisi.it>;tag=2B558754
To: <sip:201 at asb.unisi.it>;tag=as2eb08336
Call-ID: 499575437 at 192.167.125.9
CSeq: 7666 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:201 at 192.167.125.9>
Content-Type: application/sdp
Content-Length: 209

v=0
o=root 24360 24360 IN IP4 192.167.125.9
s=session
c=IN IP4 192.167.125.9
t=0 0
m=audio 18376 RTP/AVP 0 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -

 to 192.167.125.9:5062
    -- Playing 'vm-theperson' (language 'en')
moloch*CLI>

Sip read:
ACK sip:201 at 192.167.125.9 SIP/2.0
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK4236106
CSeq: 7666 ACK
To: <sip:201 at asb.unisi.it>;tag=as2eb08336
From: "203" <sip:203 at asb.unisi.it>;tag=2B558754
Call-ID: 499575437 at 192.167.125.9
Content-Length: 0
User-Agent: kphone/4.0.5
Contact: "203" <sip:203 at 192.167.125.9:5062;transport=udp>


9 headers, 0 lines
moloch*CLI>

Sip read:
BYE sip:201 at 192.167.125.9 SIP/2.0
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK1C41F388
CSeq: 7667 BYE
To: <sip:201 at asb.unisi.it>;tag=as2eb08336
From: "203" <sip:203 at asb.unisi.it>;tag=2B558754
Call-ID: 499575437 at 192.167.125.9
Content-Length: 0
User-Agent: kphone/4.0.5
Contact: "203" <sip:203 at 192.167.125.9:5062;transport=udp>


9 headers, 0 lines
Sending to 192.167.125.9 : 5062 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK1C41F388
From: "203" <sip:203 at asb.unisi.it>;tag=2B558754
To: <sip:201 at asb.unisi.it>;tag=as2eb08336
Call-ID: 499575437 at 192.167.125.9
CSeq: 7667 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:201 at 192.167.125.9>
Content-Length: 0


 to 192.167.125.9:5062
  == Spawn extension (macro-exten-vm, s, 6) exited non-zero on 'SIP/203-14e6' 
in macro 'exten-vm'
  == Spawn extension (from-internal, 201, 1) exited non-zero on 'SIP/203-14e6'
    -- Executing Macro("SIP/203-14e6", "hangupcall") in new stack
    -- Executing ResetCDR("SIP/203-14e6", "w") in new stack
    -- Executing NoCDR("SIP/203-14e6", "") in new stack
    -- Executing Wait("SIP/203-14e6", "5") in new stack
  == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 
'SIP/203-14e6' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/203-14e6'
Destroying call '499575437 at 192.167.125.9'
moloch*CLI>

Sip read:


0 headers, 0 lines
moloch*CLI>                                      
===========================8<===================================

and i get the VoiceMail apps instead of 201. Why ?

-- 
----
O-Zone ! No (C) 2005
www.zerozone.it
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