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Hachey

2003-Nov-11 03:07 UTC

[Asterisk-Users] Unable to use voicemail

Hello all.

Now I aleady installed the Asterisk.
I could make communication between 2 XLite client through Asterisk.

I tryed to test the voicemail function as follow.
 1, I make a call to 1001 from 1002
 2, Start ringing
 3, Wait untill time out for ringing

If no problem, 1001 go to voicemail and unavailable message will 
be played.
But 1001 receive a 403 forbidden massage and connection go down.
And Icould not leave a messages.
Please teach me how to resolve this problem.   

Here is configuration of Asterisk and Xlite.
#sip.conf in Asterisk
[general]
port=5060
bindaddr=0.0.0.0
nortifymimetype=text/plain
allow=all
[1001]
type=friend
username=1001
secret=1001
host=dynamic
defaultip=192.168.0.1
mailbox=1001
context=sip
canreinvite=no
[1002]
type=friend
username=1002
secret=1002
host=dynamic
defaultip=192.168.0.1
mailbox=1002
context=sip
canreinvite=no

#extensions.conf in Asterisk
[general]
static=yes
writeprotect=no
[glovals]
CONSOLE=Console/dsp
[sip]
exten => 1001,1,Dial(SIP/1001,20)
exten => 1001,2,Voicemail(u1001)
exten => 1001,102,Voicemail(b1001)
exten => 1001,103,Hungup
exten => 1002,1,Dial(SIP/1001,20)
exten => 1002,2,Voicemail(u1002)
exten => 1002,102,Voicemail(b1002)
exten => 1002,103,Hungup

#voicemail.conf in Asterisk
[local]
1001 => 1001,1001,mail address
1002 => 1002,1002,mail address

#Create mailbox by addmailbox already.

#Client configuration
User Name            1001               1002
Authorization User   same as username
PAssword             1001               1002
Domain/Realm         192.168.0.120
SIP Proxy            192.168.0.120

Here is call flow on this test.

(c)2003 Xten Networks Inc. All rights reserved.
Private build: 1008
SIP: 192.168.0.125:5061
RTP: 192.168.0.125:8000
NAT: 210.253.186.126
PXY#0: 192.168.0.120:5060

RECEIVE << 192.168.0.120:5060
NOTIFY sip:1002@192.168.0.125:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.120:5060;branch=z9hG4bK375605f3
From: "asterisk" <sip:asterisk@192.168.0.120>;tag=as633f7afa
To: <sip:1002@192.168.0.125:5061>
Contact: <sip:asterisk@192.168.0.120>
Call-ID: 6370dfe06906138479bf327d54de819c@192.168.0.120
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: text/plain
Content-Length: 36
Messages-Waiting: no
Voicemail: 0/0

SEND >> 192.168.0.120:5060
INVITE sip:1001@192.168.0.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.125:5061
From: 1002 <sip:1002@192.168.0.120:5061>;tag=337011961
To: <sip:1001@192.168.0.120>
Contact: <sip:1002@192.168.0.125:5061>
Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B@192.168.0.125
CSeq: 26502 INVITE
Content-Type: application/sdp
Content-Length: 301

v=0
o=1002 22002568 22002568 IN IP4 192.168.0.125
s=X-Lite
c=IN IP4 192.168.0.125
t=0 0
m=audio 8000 RTP/AVP 4 0 8 3 101
a=rtpmap:4 G723/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:126 x-pro-encrypted/8000

RECEIVE << 192.168.0.120:5060
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.125:5061
From: 1002 <sip:1002@192.168.0.120:5061>;tag=337011961
To: <sip:1001@192.168.0.120>;tag=as08d3281f
Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B@192.168.0.125
CSeq: 26502 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Proxy-Authenticate: Digest realm="asterisk",
nonce="05d14468"
Content-Length: 0


SEND >> 192.168.0.120:5060
ACK sip:1001@192.168.0.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.125:5061
From: 1002 <sip:1002@192.168.0.120:5061>;tag=337011961
To: <sip:1001@192.168.0.120>;tag=as08d3281f
Contact: <sip:1002@192.168.0.125:5061>
Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B@192.168.0.125
CSeq: 26502 ACK
Max-Forwards: 70
Content-Length: 0


SEND >> 192.168.0.120:5060
INVITE sip:1001@192.168.0.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.125:5061
From: 1002 <sip:1002@192.168.0.120:5061>;tag=337011961
To: <sip:1001@192.168.0.120>
Contact: <sip:1002@192.168.0.125:5061>
Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B@192.168.0.125
CSeq: 26503 INVITE
Proxy-Authorization: Digest
username="1002",realm="asterisk",nonce"05d14468",response="8fb4b56e7dae5665a8ea56a34027be5f",uri="sip:1001@192.
168.0.120"
Content-Type: application/sdp
Content-Length: 301

v=0
o=1002 22002778 22002778 IN IP4 192.168.0.125
s=X-Lite
c=IN IP4 192.168.0.125
t=0 0
m=audio 8000 RTP/AVP 4 0 8 3 101
a=rtpmap:4 G723/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:126 x-pro-encrypted/8000

RECEIVE << 192.168.0.120:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.125:5061
From: 1002 <sip:1002@192.168.0.120:5061>;tag=337011961
To: <sip:1001@192.168.0.120>;tag=as1c454920
Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B@192.168.0.125
CSeq: 26503 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1001@192.168.0.120>
Content-Length: 0


RECEIVE << 192.168.0.120:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.125:5061
From: 1002 <sip:1002@192.168.0.120:5061>;tag=337011961
To: <sip:1001@192.168.0.120>;tag=as1c454920
Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B@192.168.0.125
CSeq: 26503 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1001@192.168.0.120>
Content-Length: 0


RECEIVE << 192.168.0.120:5060
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.125:5061
From: 1002 <sip:1002@192.168.0.120:5061>;tag=337011961
To: <sip:1001@192.168.0.120>;tag=as1c454920
Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B@192.168.0.125
CSeq: 26503 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1001@192.168.0.120>
Content-Length: 0


SEND >> 192.168.0.120:5060
ACK sip:1001@192.168.0.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.125:5061
From: 1002 <sip:1002@192.168.0.120:5061>;tag=337011961
To: <sip:1001@192.168.0.120>;tag=as1c454920
Contact: <sip:1002@192.168.0.125:5061>
Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B@192.168.0.125
CSeq: 26503 ACK
Max-Forwards: 70
Content-Length: 0

Hachy

2003-Nov-11 20:32 UTC

head link

[Asterisk-Users] Re: Unable to use voicemail(Please suggestion)

Hello all

I got call log from Asterisk.
I call to ext1001 from ext1002.
But could not leave a message in the voice mail.

Please help me.

    -- Executing Dial("SIP/1002-8217", "SIP/1001|20") in new
stack
    -- Called 1001
    -- SIP/1001-25ce is ringing
    -- Nobody picked up in 20000 ms
  == Spawn extension (sip, 1001, 2) exited non-zero on 'SIP/1002-8217


>
>Hello all.
>
>Now I aleady installed the Asterisk.
>I could make communication between 2 XLite client through Asterisk.
>
>I tryed to test the voicemail function as follow.
> 1, I make a call to 1001 from 1002
> 2, Start ringing
> 3, Wait untill time out for ringing
>
>If no problem, 1001 go to voicemail and unavailable message will 
>be played.
>But 1001 receive a 403 forbidden massage and connection go down.
>And Icould not leave a messages.
>Please teach me how to resolve this problem.   
>
>Here is configuration of Asterisk and Xlite.
>#sip.conf in Asterisk
>[general]
>port=5060
>bindaddr=0.0.0.0
>nortifymimetype=text/plain
>allow=all
>[1001]
>type=friend
>username=1001
>secret=1001
>host=dynamic
>defaultip=192.168.0.1
>mailbox=1001
>context=sip
>canreinvite=no
>[1002]
>type=friend
>username=1002
>secret=1002
>host=dynamic
>defaultip=192.168.0.1
>mailbox=1002
>context=sip
>canreinvite=no
>
>#extensions.conf in Asterisk
>[general]
>static=yes
>writeprotect=no
>[glovals]
>CONSOLE=Console/dsp
>[sip]
>exten => 1001,1,Dial(SIP/1001,20)
>exten => 1001,2,Voicemail(u1001)
>exten => 1001,102,Voicemail(b1001)
>exten => 1001,103,Hungup
>exten => 1002,1,Dial(SIP/1001,20)
>exten => 1002,2,Voicemail(u1002)
>exten => 1002,102,Voicemail(b1002)
>exten => 1002,103,Hungup
>
>#voicemail.conf in Asterisk
>[local]
>1001 => 1001,1001,mail address
>1002 => 1002,1002,mail address
>
>#Create mailbox by addmailbox already.
>
>#Client configuration
>User Name            1001               1002
>Authorization User   same as username
>PAssword             1001               1002
>Domain/Realm         192.168.0.120
>SIP Proxy            192.168.0.120
>
>Here is call flow on this test.
>
>(c)2003 Xten Networks Inc. All rights reserved.
>Private build: 1008
>SIP: 192.168.0.125:5061
>RTP: 192.168.0.125:8000
>NAT: 210.253.186.126
>PXY#0: 192.168.0.120:5060
>
>RECEIVE << 192.168.0.120:5060
>NOTIFY sip:1002@192.168.0.125:5061 SIP/2.0
>Via: SIP/2.0/UDP 192.168.0.120:5060;branch=z9hG4bK375605f3
>From: "asterisk" <sip:asterisk@192.168.0.120>;tag=as633f7afa
>To: <sip:1002@192.168.0.125:5061>
>Contact: <sip:asterisk@192.168.0.120>
>Call-ID: 6370dfe06906138479bf327d54de819c@192.168.0.120
>CSeq: 102 NOTIFY
>User-Agent: Asterisk PBX
>Event: message-summary
>Content-Type: text/plain
>Content-Length: 36
>Messages-Waiting: no
>Voicemail: 0/0
>
>SEND >> 192.168.0.120:5060
>INVITE sip:1001@192.168.0.120 SIP/2.0
>Via: SIP/2.0/UDP 192.168.0.125:5061
>From: 1002 <sip:1002@192.168.0.120:5061>;tag=337011961
>To: <sip:1001@192.168.0.120>
>Contact: <sip:1002@192.168.0.125:5061>
>Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B@192.168.0.125
>CSeq: 26502 INVITE
>Content-Type: application/sdp
>Content-Length: 301
>
>v=0
>o=1002 22002568 22002568 IN IP4 192.168.0.125
>s=X-Lite
>c=IN IP4 192.168.0.125
>t=0 0
>m=audio 8000 RTP/AVP 4 0 8 3 101
>a=rtpmap:4 G723/8000
>a=rtpmap:0 pcmu/8000
>a=rtpmap:8 pcma/8000
>a=rtpmap:3 gsm/8000
>a=rtpmap:101 telephone-event/8000
>a=fmtp:101 0-15
>a=rtpmap:126 x-pro-encrypted/8000
>
>RECEIVE << 192.168.0.120:5060
>SIP/2.0 407 Proxy Authentication Required
>Via: SIP/2.0/UDP 192.168.0.125:5061
>From: 1002 <sip:1002@192.168.0.120:5061>;tag=337011961
>To: <sip:1001@192.168.0.120>;tag=as08d3281f
>Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B@192.168.0.125
>CSeq: 26502 INVITE
>User-Agent: Asterisk PBX
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>Contact: 
>Proxy-Authenticate: Digest realm="asterisk",
nonce="05d14468"
>Content-Length: 0
>
>
>SEND >> 192.168.0.120:5060
>ACK sip:1001@192.168.0.120 SIP/2.0
>Via: SIP/2.0/UDP 192.168.0.125:5061
>From: 1002 <sip:1002@192.168.0.120:5061>;tag=337011961
>To: <sip:1001@192.168.0.120>;tag=as08d3281f
>Contact: <sip:1002@192.168.0.125:5061>
>Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B@192.168.0.125
>CSeq: 26502 ACK
>Max-Forwards: 70
>Content-Length: 0
>
>
>SEND >> 192.168.0.120:5060
>INVITE sip:1001@192.168.0.120 SIP/2.0
>Via: SIP/2.0/UDP 192.168.0.125:5061
>From: 1002 <sip:1002@192.168.0.120:5061>;tag=337011961
>To: <sip:1001@192.168.0.120>
>Contact: <sip:1002@192.168.0.125:5061>
>Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B@192.168.0.125
>CSeq: 26503 INVITE
>Proxy-Authorization: Digest
username="1002",realm="asterisk",nonce>"05d14468",response="8fb4b56e7dae5665a8ea56a34027be5f",uri="sip:1001@192.
>168.0.120"
>Content-Type: application/sdp
>Content-Length: 301
>
>v=0
>o=1002 22002778 22002778 IN IP4 192.168.0.125
>s=X-Lite
>c=IN IP4 192.168.0.125
>t=0 0
>m=audio 8000 RTP/AVP 4 0 8 3 101
>a=rtpmap:4 G723/8000
>a=rtpmap:0 pcmu/8000
>a=rtpmap:8 pcma/8000
>a=rtpmap:3 gsm/8000
>a=rtpmap:101 telephone-event/8000
>a=fmtp:101 0-15
>a=rtpmap:126 x-pro-encrypted/8000
>
>RECEIVE << 192.168.0.120:5060
>SIP/2.0 100 Trying
>Via: SIP/2.0/UDP 192.168.0.125:5061
>From: 1002 <sip:1002@192.168.0.120:5061>;tag=337011961
>To: <sip:1001@192.168.0.120>;tag=as1c454920
>Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B@192.168.0.125
>CSeq: 26503 INVITE
>User-Agent: Asterisk PBX
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>Contact: <sip:1001@192.168.0.120>
>Content-Length: 0
>
>
>RECEIVE << 192.168.0.120:5060
>SIP/2.0 180 Ringing
>Via: SIP/2.0/UDP 192.168.0.125:5061
>From: 1002 <sip:1002@192.168.0.120:5061>;tag=337011961
>To: <sip:1001@192.168.0.120>;tag=as1c454920
>Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B@192.168.0.125
>CSeq: 26503 INVITE
>User-Agent: Asterisk PBX
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>Contact: <sip:1001@192.168.0.120>
>Content-Length: 0
>
>
>RECEIVE << 192.168.0.120:5060
>SIP/2.0 403 Forbidden
>Via: SIP/2.0/UDP 192.168.0.125:5061
>From: 1002 <sip:1002@192.168.0.120:5061>;tag=337011961
>To: <sip:1001@192.168.0.120>;tag=as1c454920
>Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B@192.168.0.125
>CSeq: 26503 INVITE
>User-Agent: Asterisk PBX
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>Contact: <sip:1001@192.168.0.120>
>Content-Length: 0
>
>
>SEND >> 192.168.0.120:5060
>ACK sip:1001@192.168.0.120 SIP/2.0
>Via: SIP/2.0/UDP 192.168.0.125:5061
>From: 1002 <sip:1002@192.168.0.120:5061>;tag=337011961
>To: <sip:1001@192.168.0.120>;tag=as1c454920
>Contact: <sip:1002@192.168.0.125:5061>
>Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B@192.168.0.125
>CSeq: 26503 ACK
>Max-Forwards: 70
>Content-Length: 0

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