[Asterisk-Users] Re: Unable to use voicemail(Please suggestion)
CW_ASN - Gus
cw_asn at fibertel.com.ar
Mon Nov 12 08:31:13 CST 2007
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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