[Asterisk-Users] HELP ME!!!! Asterisk don't do calls
Michele "O-Zone" Pinassi
zerozone.liste at gmail.com
Wed May 18 03:31:14 MST 2005
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 at asb.unisi.it SIP/2.0
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK64FE3538
CSeq: 1114 INVITE
To: <sip:201 at asb.unisi.it>
Content-Type: application/sdp
From: "203" <sip:203 at asb.unisi.it>;tag=1CE28F8
Call-ID: 1646594512 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 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 at asb.unisi.it>;tag=1CE28F8
To: <sip:201 at asb.unisi.it>;tag=as3c1a1273
Call-ID: 1646594512 at 192.167.125.9
CSeq: 1114 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="0ae53906"
Content-Length: 0
to 192.167.125.9:5062
Scheduling destruction of call '1646594512 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=z9hG4bK64FE3538
CSeq: 1114 ACK
To: <sip:201 at asb.unisi.it>;tag=as3c1a1273
From: "203" <sip:203 at asb.unisi.it>;tag=1CE28F8
Call-ID: 1646594512 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=z9hG4bK7602DA72
CSeq: 1115 INVITE
To: <sip:201 at asb.unisi.it>
Proxy-Authorization: Digest username="203", realm="asterisk", nonce="0ae53906", uri="sip:201 at asb.unisi.it", cnonce="abcdefghi", nc=00000001, response="58e82c67b3c712ffb39220e473903007", opaque="", algorithm="MD5"
Content-Type: application/sdp
From: "203" <sip:203 at asb.unisi.it>;tag=1CE28F8
Call-ID: 1646594512 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 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 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=z9hG4bK7602DA72
From: "203" <sip:203 at asb.unisi.it>;tag=1CE28F8
To: <sip:201 at asb.unisi.it>
Call-ID: 1646594512 at 192.167.125.9
CSeq: 1115 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-f9ee", "exten-vm|201 at 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 at 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 at asb.unisi.it>;tag=1CE28F8
To: <sip:201 at asb.unisi.it>;tag=as50e9a0f8
Call-ID: 1646594512 at 192.167.125.9
CSeq: 1115 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 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 at 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 at asb.unisi.it>;tag=as50e9a0f8
From: "203" <sip:203 at asb.unisi.it>;tag=1CE28F8
Call-ID: 1646594512 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
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/0' (language 'en')
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=z9hG4bK6272BA5A
CSeq: 1116 BYE
To: <sip:201 at asb.unisi.it>;tag=as50e9a0f8
From: "203" <sip:203 at asb.unisi.it>;tag=1CE28F8
Call-ID: 1646594512 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=z9hG4bK6272BA5A
From: "203" <sip:203 at asb.unisi.it>;tag=1CE28F8
To: <sip:201 at asb.unisi.it>;tag=as50e9a0f8
Call-ID: 1646594512 at 192.167.125.9
CSeq: 1116 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-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 at default,201)
exten => 202,1,Macro(exten-vm,202 at default,202)
exten => 203,1,Macro(exten-vm,203 at default,203)
exten => 204,1,Macro(exten-vm,204 at 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
seret=
qualify=200
port=5060
pickupgroup=
nat=never
mailbox=20x at default
host=dynamic
dtmfmode=rfc2833
disallow=
context=from-internal
canreinvite=no
callgroup=
callerid="djdjdj" <20x>
allow=
Help !!!!!!!!!!!
--
----
O-Zone ! No (C) 2005
www.zerozone.it
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