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@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@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@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@192.168.182.10>;tag=as2a80256e
To:
<sip:200@192.168.182.10:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml
Contact:
<sip:7005551212@192.168.182.10>
Call-ID: 7b13de3e4550bb8d132cb9e95f004ae4@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@192.168.182.10>;tag=as2a80256e
To:
<sip:200@192.168.182.10:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml
Call-Id:
7b13de3e4550bb8d132cb9e95f004ae4@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@sipXvxml
dev*CLI>
Sip read:
SIP/2.0 180 Ringing
From: "Not Available"
<sip:7005551212@192.168.182.10>;tag=as2a80256e
To:
<sip:200@192.168.182.10:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml;tag=1134035663
Call-Id:
7b13de3e4550bb8d132cb9e95f004ae4@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@192.168.182.10>;tag=as2a80256e
To:
<sip:200@192.168.182.10:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml;tag=1134035663
Call-Id:
7b13de3e4550bb8d132cb9e95f004ae4@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@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@192.168.182.10>;tag=as2a80256e
To:
<sip:200@192.168.182.10:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml;tag=1134035663
Contact: <sip:7005551212@192.168.182.10>
Call-ID: 7b13de3e4550bb8d132cb9e95f004ae4@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@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
This looks really interesting and opens up a number of possible end user
solutions if you can get it working.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
c.lacetera@tin.it
Sent: Tuesday, October 26, 2004 8:19 AM
To: asterisk-users@lists.digium.com
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/200@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@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@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@192.168.182.10>;tag=as2a80256e
To:
<sip:200@192.168.182.10:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex
.vxml
Contact:
<sip:7005551212@192.168.182.10>
Call-ID: 7b13de3e4550bb8d132cb9e95f004ae4@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@192.168.182.10>;tag=as2a80256e
To:
<sip:200@192.168.182.10:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex
.vxml
Call-Id:
7b13de3e4550bb8d132cb9e95f004ae4@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@sipXvxml
dev*CLI>
Sip read:
SIP/2.0 180 Ringing
From: "Not Available"
<sip:7005551212@192.168.182.10>;tag=as2a80256e
To:
<sip:200@192.168.182.10:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex
.vxml;tag=1134035663
Call-Id:
7b13de3e4550bb8d132cb9e95f004ae4@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@192.168.182.10>;tag=as2a80256e
To:
<sip:200@192.168.182.10:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex
.vxml;tag=1134035663
Call-Id:
7b13de3e4550bb8d132cb9e95f004ae4@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@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@192.168.182.10>;tag=as2a80256e
To:
<sip:200@192.168.182.10:5100>;play=http%3A%2F%2Flocalhost%2Fvxml%2Findex
.vxml;tag=1134035663
Contact: <sip:7005551212@192.168.182.10>
Call-ID: 7b13de3e4550bb8d132cb9e95f004ae4@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@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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Hi All, I searched the mail archives, and found a few older posts about people interested in using asterisk as a telephony platform for VoiceXML using OpenVXI or other browsers. Has anyone been successful with any sort of intigration ? I have looked at trying to integrate OpenVXI with other telephoney platforms in the past, and looking at how it can be done with asterisk, looks like a focused way to dive into many aspects of asterisk. If anyone can share their experiences either on or off list, it would be appriciated. Thanks Shahed