Hello,
I've developed a forwarding application using VXML and TCL, but I'm having a very strange and serious problem when playing an audio file first and then forwarding a call to landline phone.
Below a summary of the different scenarios
| ID | Source | Destination | CostFile | Result |
| 1 | LANDLINE | MOBILE | YES | OK |
| 2 | LANDLINE | MOBILE | NO | OK |
| 3 | LANDLINE | LANDLINE | YES | NA |
| 4 | LANDLINE | LANDLINE | NO | NA |
| 5 | MOBILE | LANDLINE | YES | NOK |
| 6 | MOBILE | LANDLINE | NO | OK |
| 7 | MOBILE | MOBILE | YES | OK |
| 8 | MOBILE | MOBILE | NO | OK |
As you can see almost all other scenarios work just fine. What happens when playing an audio file first and then transferring to a landline phone is that the audio file is played but then the call leg between the calleer and our gateway is disconnected but the gateway still initiates a call with the destination, the destination rings. I'm not using the standard <transfer> element, but a custom one written for CVP which uses a custom TCL script. This can be found here:https://developer.cisco.com/web/cvp/forums//message_boards/view_message/2685049_19_delta=20&_19_keywords=&_19_advancedSearch=false&_19_andOperator=true&cur=2
Edit: I've tested the same scenario with the standard <transfer> element for VXML and it's the same result. So I'm guessing it's not the fault of the tcl script.
Landline destination is ISDN.
When the caller calls our gateway but is NOT played an audio file and then is forwarded to a landline works just fine.
I compared the SIP messages between scenario 5 and 6. The SIP messages start to be different here:
SCENARIO 5:
Received:
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK1FFC2
To: "xxxxx" <sip:xxxxxxx@x.x.x.x>;tag=3535282942-113333
From: <sip:xxxxx@x.x.x.x:5060>;tag=4356C-84D
Call-ID: 802674-3535282942-113330@xxxxx.xxxxx.local
CSeq: 102 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Call-Info: <sip:x.x.x.x>;method="NOTIFY;Event=telephone-event;Duration=1000"
Content-Length: 0
Scenario 6:
Received:
SIP/2.0 200 OK
Session-Expires: 3600;refresher=uas
Require: timer
Via: SIP/2.0/UDP x.x.x.x5060;branch=z9hG4bKD2538
To: "xxxxx" <sip:xxxxxx@x.x.x.x.x>;tag=3535282801-21158
From: <sip:xxxxx@x.x.x.x:5060>;tag=20E38-E18
Remote-Party-Id: <sip:xxxxx@x.x.x.x>;screen=yes;privacy=off
Call-ID: 802397-3535282801-21150@xxxxx.xxxxx.local
CSeq: 101 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Contact: <sip:xxxxx@x.x.x.x:5060>
Call-Info: <sip:x.x.x.x>;method="NOTIFY;Event=telephone-event;Duration=1000"
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 209
v=0
o=AM00SBC03 7663 5850 IN IP4 x.x.x.x
s=sip call
c=IN IP4 x.x.x.x
t=0 0
m=audio 20842 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
Does anyone know why this is happening?
Grant
