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

Hachy hashimoto at bmc-j.net
Tue Nov 11 20:32:38 MST 2003 

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 at 192.168.0.125:5061 SIP/2.0
>Via: SIP/2.0/UDP 192.168.0.120:5060;branch=z9hG4bK375605f3
>From: "asterisk" <sip:asterisk at 192.168.0.120>;tag=as633f7afa
>To: <sip:1002 at 192.168.0.125:5061>
>Contact: <sip:asterisk at 192.168.0.120>
>Call-ID: 6370dfe06906138479bf327d54de819c at 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 at 192.168.0.120 SIP/2.0
>Via: SIP/2.0/UDP 192.168.0.125:5061
>From: 1002 <sip:1002 at 192.168.0.120:5061>;tag=337011961
>To: <sip:1001 at 192.168.0.120>
>Contact: <sip:1002 at 192.168.0.125:5061>
>Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B at 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 at 192.168.0.120:5061>;tag=337011961
>To: <sip:1001 at 192.168.0.120>;tag=as08d3281f
>Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B at 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 at 192.168.0.120 SIP/2.0
>Via: SIP/2.0/UDP 192.168.0.125:5061
>From: 1002 <sip:1002 at 192.168.0.120:5061>;tag=337011961
>To: <sip:1001 at 192.168.0.120>;tag=as08d3281f
>Contact: <sip:1002 at 192.168.0.125:5061>
>Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B at 192.168.0.125
>CSeq: 26502 ACK
>Max-Forwards: 70
>Content-Length: 0
>
>
>SEND >> 192.168.0.120:5060
>INVITE sip:1001 at 192.168.0.120 SIP/2.0
>Via: SIP/2.0/UDP 192.168.0.125:5061
>From: 1002 <sip:1002 at 192.168.0.120:5061>;tag=337011961
>To: <sip:1001 at 192.168.0.120>
>Contact: <sip:1002 at 192.168.0.125:5061>
>Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B at 192.168.0.125
>CSeq: 26503 INVITE
>Proxy-Authorization: Digest username="1002",realm="asterisk",nonce=
>"05d14468",response="8fb4b56e7dae5665a8ea56a34027be5f",uri="sip:1001 at 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 at 192.168.0.120:5061>;tag=337011961
>To: <sip:1001 at 192.168.0.120>;tag=as1c454920
>Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B at 192.168.0.125
>CSeq: 26503 INVITE
>User-Agent: Asterisk PBX
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>Contact: <sip:1001 at 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 at 192.168.0.120:5061>;tag=337011961
>To: <sip:1001 at 192.168.0.120>;tag=as1c454920
>Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B at 192.168.0.125
>CSeq: 26503 INVITE
>User-Agent: Asterisk PBX
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>Contact: <sip:1001 at 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 at 192.168.0.120:5061>;tag=337011961
>To: <sip:1001 at 192.168.0.120>;tag=as1c454920
>Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B at 192.168.0.125
>CSeq: 26503 INVITE
>User-Agent: Asterisk PBX
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>Contact: <sip:1001 at 192.168.0.120>
>Content-Length: 0
>
>
>SEND >> 192.168.0.120:5060
>ACK sip:1001 at 192.168.0.120 SIP/2.0
>Via: SIP/2.0/UDP 192.168.0.125:5061
>From: 1002 <sip:1002 at 192.168.0.120:5061>;tag=337011961
>To: <sip:1001 at 192.168.0.120>;tag=as1c454920
>Contact: <sip:1002 at 192.168.0.125:5061>
>Call-ID: 1A20F406-F972-4151-8375-F6B3C079943B at 192.168.0.125
>CSeq: 26503 ACK
>Max-Forwards: 70
>Content-Length: 0

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