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Michele "O-Zone" Pinassi

2005-May-17 03:58 UTC

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

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@asb.unisi.it SIP/2.0
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK78ED0E83
CSeq: 7665 INVITE
To: <sip:201@asb.unisi.it>
Content-Type: application/sdp
From: "203" <sip:203@asb.unisi.it>;tag=2B558754
Call-ID: 499575437@192.167.125.9
Subject: sip:203@asb.unisi.it
Content-Length: 187
User-Agent: kphone/4.0.5
Contact: "203" <sip:203@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@asb.unisi.it>;tag=2B558754
To: <sip:201@asb.unisi.it>;tag=as17f37979
Call-ID: 499575437@192.167.125.9
CSeq: 7665 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:201@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@192.167.125.9' in 15000 ms
Found user '203'
moloch*CLI>

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


9 headers, 0 lines
moloch*CLI>

Sip read:
INVITE sip:201@asb.unisi.it SIP/2.0
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK4236106
CSeq: 7666 INVITE
To: <sip:201@asb.unisi.it>
Proxy-Authorization: Digest username="203",
realm="asterisk",
nonce="2149fad7", uri="sip:201@asb.unisi.it",
cnonce="abcdefghi",
nc=00000001, response="b1a9c4ee2ac7065635f681a281dcec25",
opaque="",
algorithm="MD5"
Content-Type: application/sdp
From: "203" <sip:203@asb.unisi.it>;tag=2B558754
Call-ID: 499575437@192.167.125.9
Subject: sip:203@asb.unisi.it
Content-Length: 187
User-Agent: kphone/4.0.5
Contact: "203" <sip:203@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@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@asb.unisi.it>;tag=2B558754
To: <sip:201@asb.unisi.it>
Call-ID: 499575437@192.167.125.9
CSeq: 7666 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:201@192.167.125.9>
Content-Length: 0


 to 192.167.125.9:5062
    -- Executing Macro("SIP/203-14e6",
"exten-vm|201@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@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@asb.unisi.it>;tag=2B558754
To: <sip:201@asb.unisi.it>;tag=as2eb08336
Call-ID: 499575437@192.167.125.9
CSeq: 7666 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:201@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@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@asb.unisi.it>;tag=as2eb08336
From: "203" <sip:203@asb.unisi.it>;tag=2B558754
Call-ID: 499575437@192.167.125.9
Content-Length: 0
User-Agent: kphone/4.0.5
Contact: "203" <sip:203@192.167.125.9:5062;transport=udp>


9 headers, 0 lines
moloch*CLI>

Sip read:
BYE sip:201@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@asb.unisi.it>;tag=as2eb08336
From: "203" <sip:203@asb.unisi.it>;tag=2B558754
Call-ID: 499575437@192.167.125.9
Content-Length: 0
User-Agent: kphone/4.0.5
Contact: "203" <sip:203@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@asb.unisi.it>;tag=2B558754
To: <sip:201@asb.unisi.it>;tag=as2eb08336
Call-ID: 499575437@192.167.125.9
CSeq: 7667 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:201@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@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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Eric Wieling aka ManxPower

2005-May-18 06:08 UTC

head link

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

Michele "O-Zone" Pinassi wrote:
> 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>    
"sip show registry" shows remote systems Asterisk is registered to.

Try "sip show peers"

-- 
Always do right. This will gratify some people and astonish the rest.
Mark Twain

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