This looks really interesting and opens up a number of possible end user solutions if you can get it working.
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, October 26, 2004 8:19 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ASTERISK and VoiceXML 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/[EMAIL PROTECTED],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 [EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5100 SIP/2.0 Via: SIP/2.0/UDP 192.168.182.10:5060;branch=z9hG4bK6baf7728 From: "Not Available" <sip:[EMAIL PROTECTED]>;tag=as2a80256e To: <sip:[EMAIL PROTECTED]:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex .vxml Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;tag=as2a80256e To: <sip:[EMAIL PROTECTED]:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex .vxml Call-Id: [EMAIL PROTECTED] Cseq: 102 INVITE Via: SIP/2.0/UDP 192.168.182.10:5060;branch=z9hG4bK6baf7728 Content-Length: 0 7 headers, 0 lines -- Called [EMAIL PROTECTED] dev*CLI> Sip read: SIP/2.0 180 Ringing From: "Not Available" <sip:[EMAIL PROTECTED]>;tag=as2a80256e To: <sip:[EMAIL PROTECTED]:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex .vxml;tag=1134035663 Call-Id: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;tag=as2a80256e To: <sip:[EMAIL PROTECTED]:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex .vxml;tag=1134035663 Call-Id: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5100 SIP/2.0 Via: SIP/2.0/UDP 192.168.182.10:5060;branch=z9hG4bK00e223fc From: "Not Available" <sip:[EMAIL PROTECTED]>;tag=as2a80256e To: <sip:[EMAIL PROTECTED]:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex .vxml;tag=1134035663 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 ________________________________________________ dev*CLI> sip show channel 3f4ba1f2533 dev*CLI> * SIP Call Direction: Outgoing Call-ID: [EMAIL PROTECTED] 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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
