[Asterisk-Users] Re: Unable to use voicemail(Thanks)
BMC Hashimoto
hashimoto at bmc-j.net
Wed Nov 12 17:58:38 MST 2003
Thank you Gus
I found some mistake in extension.conf
exten => 1001,2.Voicemail(u1001)
This must be change to ",2,Voicemail(----".
Now I use some hard phone, so I would better try another codec.
Thanks for your good advice
>Try with another codec different than G.723. Use GSM o G.711 for this.
>You could disable G.723 in your sip.conf
>
>disallow=all
>allow=gsm
>allow=alaw
>allow=ulaw
>
>Hope this helps,
>
>Gus
>
>----- Original Message -----
>From: "Hachy" <hashimoto at bmc-j.net>
>To: <asterisk-users at lists.digium.com>
>Sent: Wednesday, November 12, 2003 12:32 AM
>Subject: [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 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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>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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Broadband Mobile Communications INC.
System Integration Devision
Yukiyasu Hashimoto
32F Izumi Girden Tower 1-6-1 Roppongi
Minatoku Tokyo Japan
TEL/FAX:+813-6733-3045/+813-5114-3155
Mail:hashimoto at bmc-j.net
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