[Asterisk-Users] ASTERISK and VoiceXML

c.lacetera at tin.it c.lacetera at tin.it
Tue Oct 26 05:19:03 MST 2004


Hi to all

There's a intersting project  http://www.sipfoundry.org/sipXvxml/index.html
of a sip PBX that use a VXML gateway for voice mail

This part maybe a standalone, will' be intersting if the two projects join
effort for make VXML Interaction in asterisk.


I tryed to compile and execute this sw. But i have some trouble to make
it work:

there are my config

extensions.conf
exten => _1.,1,Answer
exten => _1.,2,SetVar(VXML_URL=play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml)
exten => _1.,3,Dial(SIP/200 at sipXvxml,30,t)
exten => _1.,4,hangup

sip.conf
[sipXvxml]
type=friend
insecure=yes
username=100
reinvite=no
host=192.168.182.10
port=5100
disallow=all
allow=alaw
nat=no

The sipXvxml answer at call but seem to remain appended....

last sipXvxml log is:
MpCallFlowGraph::synchronize()
RECEIVING RTP
Call-19 SIP ACK method received
No SDP in message
Connection state change - isLocal 0
         for call Call-19
         with callid 07e524d325ab61be37b028576fe91ba4 at 192.168.182.10
         from: CONNECTION_ESTABLISHED
         to CONNECTION_ESTABLISHED
         (cause=0) is not allowed.


Someone had played with this ? 
Some Suggestions ???

Regards


-----Asterisk SIP LOG-------------

Answering with capability 0x8(ALAW)                                    
                                                     Answering with non-codec
capability 0x1(G723)                                                   
                            12 headers, 10 lines                       
                                                                       
         Reliably Transmitting:                                        
                                                              INVITE sip:200 at 192.168.182.10:5100
SIP/2.0                                                                
                  Via: SIP/2.0/UDP 192.168.182.10:5060;branch=z9hG4bK6baf7728
                                                                 From: "Not
Available" <sip:7005551212 at 192.168.182.10>;tag=as2a80256e              
                                          To: <sip:200 at 192.168.182.10:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml
                                           Contact: <sip:7005551212 at 192.168.182.10>
                                                                       
            Call-ID: 7b13de3e4550bb8d132cb9e95f004ae4 at 192.168.182.10   
                                                                 CSeq: 102
INVITE                                                                 
                                           User-Agent: Asterisk PBX    
                                                                       
                        Date: Tue, 26 Oct 2004 01:59:57 GMT            
                                                                       
     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER                   
                                                          Content-Type:
application/sdp                                                        
                                       Content-Length: 220             
                                                                       
                                                                       
                                                                       
 v=0                                                                   
                                                      o=root 30121 30121
IN IP4 192.168.182.10                                                  
                                  s=session                            
                                                                       
               c=IN IP4 192.168.182.10                                 
                                                                    t=0
0                                                                      
                                                 m=audio 18946 RTP/AVP 8
101                                                                    
                             a=rtpmap:8 PCMA/8000                      
                                                                       
          a=rtpmap:101 telephone-event/8000                            
                                                               a=fmtp:101
0-16                                                                   
                                          a=silenceSupp:off - - - -    
                                                                       
                        (no NAT) to 192.168.182.10:5100                
                                                                       
    dev*CLI>                                                           
                                                                       
                                                                       
                                      Sip read:                        
                                                                       
                   SIP/2.0 100 Trying                                  
                                                                       
From: "Not Available" <sip:7005551212 at 192.168.182.10>;tag=as2a80256e   
                                                     To: <sip:200 at 192.168.182.10:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml
                                           Call-Id: 7b13de3e4550bb8d132cb9e95f004ae4 at 192.168.182.10
                                                                    Cseq:
102 INVITE                                                             
                                               Via: SIP/2.0/UDP 192.168.182.10:5060;branch=z9hG4bK6baf7728
                                                                 Content-Length:
0

7 headers, 0 lines                                                     
                                                         -- Called 200 at sipXvxml
                                                                       
                          dev*CLI>                                     
                                                                       
                                                                       
                                                            Sip read:  
                                                                       
                                         SIP/2.0 180 Ringing           
                                                                       
                      From: "Not Available" <sip:7005551212 at 192.168.182.10>;tag=as2a80256e
                                                        To: <sip:200 at 192.168.182.10:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml;tag=1134035663
                            Call-Id: 7b13de3e4550bb8d132cb9e95f004ae4 at 192.168.182.10
                                                                    Cseq:
102 INVITE                                                             
                                               Via: SIP/2.0/UDP 192.168.182.10:5060;branch=z9hG4bK6baf7728
                                                                 Date: Tue,
26 Oct 2004 01:59:57 GMT                                               
                                          Contact: sip:192.168.182.10:5100
                                                                       
                    User-Agent: sipX/2.6.0 (Linux)                     
                                                                       
 Accept-Language: en                                                   
                                                      Allow: INVITE, ACK,
CANCEL, BYE, REFER, OPTIONS, NOTIFY                                    
                                 Supported: sip-cc, sip-cc-01, replaces
                                                                       
              Content-Length: 0


Sip read:                                                              
                                                     SIP/2.0 200 OK    
                                                                       
                                  From: "Not Available" <sip:7005551212 at 192.168.182.10>;tag=as2a80256e
                                                        To: <sip:200 at 192.168.182.10:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml;tag=1134035663
                            Call-Id: 7b13de3e4550bb8d132cb9e95f004ae4 at 192.168.182.10
                                                                    Cseq:
102 INVITE                                                             
                                               Content-Type: application/sdp
                                                                       
                       Content-Length: 177                             
                                                                       
    Via: SIP/2.0/UDP 192.168.182.10:5060;branch=z9hG4bK6baf7728        
                                                         Date: Tue, 26 Oct
2004 01:59:57 GMT                                                      
                                   Contact: sip:192.168.182.10:5100    
                                                                       
                Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY
                                                                     User-Agent:
sipX/2.6.0 (Linux)                                                     
                                         Accept-Language: en           
                                                                       
                      Supported: sip-cc, sip-cc-01, replaces           
                                                                       
                                                                       
                                                        v=0            
                                                                       
                                     s=phone-call                      
                                                                       
                  o=Pingtel 5 5 IN IP4 192.168.182.10                  
                                                                       c=IN
IP4 192.168.182.10                                                     
                                                t=0 0                  
                                                                       
                             m=audio 9000 RTP/AVP 8 101                
                                                                       
          a=rtpmap:8 pcma/8000/1                                       
                                                               a=rtpmap:101
telephone-event/8000/1                                                 
                                                                       
                                                                       
                     14 headers, 8 lines                               
                                                                       
  Found RTP audio format 8                                             
                                                       Found RTP audio format
101                                                                    
                              Peer audio RTP is at port 192.168.182.10:9000
                                                                       
       Found description format pcma                                   
                                                            Found description
format telephone-event                                                 
                                   Capabilities: us - 0x8(ALAW), peer -
audio=0x8(ALAW)/video=0x0(EMPTY), combined - 0x8(ALAW)                 
                Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723),
combined - 0x1(G723)                                               list_route:
hop: <sip:192.168.182.10:5100>                                         
                                         set_destination: Parsing <sip:192.168.182.10:5100>
for address/port to send to
set_destination: set destination to 192.168.182.10, port 5100
Transmitting:
ACK sip:200 at 192.168.182.10:5100 SIP/2.0
Via: SIP/2.0/UDP 192.168.182.10:5060;branch=z9hG4bK00e223fc
From: "Not Available" <sip:7005551212 at 192.168.182.10>;tag=as2a80256e
To: <sip:200 at 192.168.182.10:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml;tag=1134035663
Contact: <sip:7005551212 at 192.168.182.10>
Call-ID: 7b13de3e4550bb8d132cb9e95f004ae4 at 192.168.182.10
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

________________________________________________


dev*CLI> sip show channel 3f4ba1f2533
dev*CLI>
  * SIP Call
  Direction:              Outgoing
  Call-ID:                3f4ba1f253322dc9446ea8f523eef9da at 192.168.182.10
  Our Codec Capability:   8
  Non-Codec Capability:   1
  Their Codec Capability:   8
  Joint Codec Capability:   8
  Format                  ALAW
  Theoretical Address:    192.168.182.10:5100
  Received Address:       192.168.182.10:32787
  NAT Support:            No
  Our Tag:                1913898535
  Their Tag:              1473482055
  SIP User agent:         sipX/2.6.0 (Linux)
  Username:               200
  Peername:               100
  Original uri:           sip:192.168.182.10:5100
  Need Destroy:           0
  Last Message:           Tx: ACK
  Promiscuous Redir:      No
  Route:                  sip:192.168.182.10:5100
  DTMF Mode:              rfc2833






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