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New Member

Call leg disconnect during transfer

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

IDSourceDestinationCostFileResult
1LANDLINEMOBILEYESOK
2LANDLINEMOBILENOOK
3LANDLINELANDLINEYESNA
4LANDLINELANDLINENONA
5MOBILELANDLINEYESNOK
6MOBILELANDLINENOOK
7MOBILEMOBILEYESOK
8MOBILEMOBILENOOK


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

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New Member

Call leg disconnect during transfer

Can someone please shed some light on this problem? I'm kinda stuck on this.

Thanks,

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