Michele "O-Zone" Pinassi
2005-May-18 03:31 UTC
[Asterisk-Users] HELP ME!!!! Asterisk don't do calls
Hi all,
as in last mail, i've installed Asterisk from CVS and AMP to manage it.
I've made 4 extensions:
moloch*CLI> sip show peers
Name/username Host Dyn Nat ACL Mask Port Status
204/204 (Unspecified) D 255.255.255.255 0 UNKNOWN
203/203 192.167.125.9 D 255.255.255.255 5062 OK (3 ms)
202/202 (Unspecified) D 255.255.255.255 0 UNKNOWN
201/201 192.167.125.12 D 255.255.255.255 5060 OK (3 ms)
moloch*CLI>
as you can see, 201 and 203 are on-line but, if i call from 203 to 201, i
immediately go to voicemail instead of doing call to 201. Here's the SIP
call flow:
moloch*CLI>
Sip read:
INVITE sip:201@asb.unisi.it SIP/2.0
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK64FE3538
CSeq: 1114 INVITE
To: <sip:201@asb.unisi.it>
Content-Type: application/sdp
From: "203" <sip:203@asb.unisi.it>;tag=1CE28F8
Call-ID: 1646594512@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 36808 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=z9hG4bK64FE3538
From: "203" <sip:203@asb.unisi.it>;tag=1CE28F8
To: <sip:201@asb.unisi.it>;tag=as3c1a1273
Call-ID: 1646594512@192.167.125.9
CSeq: 1114 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="0ae53906"
Content-Length: 0
to 192.167.125.9:5062
Scheduling destruction of call '1646594512@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=z9hG4bK64FE3538
CSeq: 1114 ACK
To: <sip:201@asb.unisi.it>;tag=as3c1a1273
From: "203" <sip:203@asb.unisi.it>;tag=1CE28F8
Call-ID: 1646594512@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=z9hG4bK7602DA72
CSeq: 1115 INVITE
To: <sip:201@asb.unisi.it>
Proxy-Authorization: Digest username="203",
realm="asterisk", nonce="0ae53906",
uri="sip:201@asb.unisi.it", cnonce="abcdefghi", nc=00000001,
response="58e82c67b3c712ffb39220e473903007", opaque="",
algorithm="MD5"
Content-Type: application/sdp
From: "203" <sip:203@asb.unisi.it>;tag=1CE28F8
Call-ID: 1646594512@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 36808 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 192.167.125.9:36808
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=z9hG4bK7602DA72
From: "203" <sip:203@asb.unisi.it>;tag=1CE28F8
To: <sip:201@asb.unisi.it>
Call-ID: 1646594512@192.167.125.9
CSeq: 1115 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-f9ee",
"exten-vm|201@default|201") in new stack
-- Executing SetVar("SIP/203-f9ee",
"FROMCONTEXT=exten-vm") in new stack
-- Executing GotoIf("SIP/203-f9ee", "0?novm|1:3") in new
stack
-- Goto (macro-exten-vm,s,3)
-- Executing GotoIf("SIP/203-f9ee", "0?novm|1") in new
stack
-- Executing Macro("SIP/203-f9ee", "dial|30|tr|201") in
new stack
-- Executing AGI("SIP/203-f9ee", "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-f9ee", "1") in new stack
-- Executing VoiceMail("SIP/203-f9ee", "u201@default")
in new stack
We're at 192.167.125.9 port 15724
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=z9hG4bK7602DA72
From: "203" <sip:203@asb.unisi.it>;tag=1CE28F8
To: <sip:201@asb.unisi.it>;tag=as50e9a0f8
Call-ID: 1646594512@192.167.125.9
CSeq: 1115 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 29772 29772 IN IP4 192.167.125.9
s=session
c=IN IP4 192.167.125.9
t=0 0
m=audio 15724 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
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=z9hG4bK7602DA72
CSeq: 1115 ACK
To: <sip:201@asb.unisi.it>;tag=as50e9a0f8
From: "203" <sip:203@asb.unisi.it>;tag=1CE28F8
Call-ID: 1646594512@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
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/0' (language 'en')
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=z9hG4bK6272BA5A
CSeq: 1116 BYE
To: <sip:201@asb.unisi.it>;tag=as50e9a0f8
From: "203" <sip:203@asb.unisi.it>;tag=1CE28F8
Call-ID: 1646594512@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=z9hG4bK6272BA5A
From: "203" <sip:203@asb.unisi.it>;tag=1CE28F8
To: <sip:201@asb.unisi.it>;tag=as50e9a0f8
Call-ID: 1646594512@192.167.125.9
CSeq: 1116 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-f9ee' in macro 'exten-vm'
== Spawn extension (from-internal, 201, 1) exited non-zero on
'SIP/203-f9ee'
-- Executing Macro("SIP/203-f9ee", "hangupcall") in new
stack
-- Executing ResetCDR("SIP/203-f9ee", "w") in new stack
-- Executing NoCDR("SIP/203-f9ee", "") in new stack
-- Executing Wait("SIP/203-f9ee", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/203-f9ee' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/203-f9ee'
moloch*CLI>
and this is the extensions definitions:
[ext-local]
include => ext-local-custom
exten => 201,1,Macro(exten-vm,201@default,201)
exten => 202,1,Macro(exten-vm,202@default,202)
exten => 203,1,Macro(exten-vm,203@default,203)
exten => 204,1,Macro(exten-vm,204@default,204)
; Ring an extension, if the extension is busy or there is no answer send it
; to voicemail
; ARGS: $VMBOX, $EXT
[macro-exten-vm]
exten => s,1,Setvar(FROMCONTEXT=exten-vm)
exten => s,2,GotoIf($[${CHANNEL:0:5} = Local]?novm,1:3) ; if the channel is
Local, then do not go to voicemail. This is $
exten => s,3,GotoIf($[${ARG1} = novm]?novm,1)
exten => s,4,Macro(dial,${RINGTIMER},${DIAL_OPTIONS},${ARG2})
exten => s,5,Wait(1)
exten => s,6,Voicemail(u${ARG1}) ; no answer to voicemail
exten => s,7,Macro(hangupcall)
exten => s,106,Wait(1)
exten => s,107,Voicemail(b${ARG1})
exten => o,1,Background(one-moment-please) ; 0 during vm message will
hangup
exten => o,2,goto(from-pstn,s,1)
exten => a,1,Goto(app-directory,*411,1)
exten => a,2,Hangup
exten => novm,1,Macro(dial,120,${DIAL_OPTIONS},${ARG2})
exten => novm,2,Wait(1)
exten => novm,3,Playback(vm-nobodyavail)
exten => novm,4,Playback(allison7/pls-try-call-later)
exten => novm,5,Hangup
there's the extension definitions (the same for 201,202,203,204):
[20x]
username=20x
type=friend
seretqualify=200
port=5060
pickupgroupnat=never
mailbox=20x@default
host=dynamic
dtmfmode=rfc2833
disallowcontext=from-internal
canreinvite=no
callgroupcallerid="djdjdj" <20x>
allow
Help !!!!!!!!!!!
--
----
O-Zone ! No (C) 2005
www.zerozone.it
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