Problem with Transfer Element for performing Bridge transfer ?

[maryamsharifi]

Hello,

After changing the GateWay configuration on the Pre-Production environment We have some issue for transferring the call to the external phone number by CVP:

We have 2 GateWay ( one is connected to T2, another one only VXML), there is a Call Studio application for transferring the

calls to the external phone number, in this application we used Transfer Element with these configurations :
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<!DOCTYPE configuration SYSTEM "../../../../dtds/VoiceElementConfiguration.dtd">
<configuration class="com.audium.server.voiceElement.transfer.MTransfer" serial="0000">
<setting name="transfer_destination">{0}</setting>
<setting name="destination_type">tel</setting>
<setting name="connect_timeout">60s</setting>
<setting name="max_transfer_time">0s</setting>
<setting name="bridge">true</setting>
<substitute index="0">
<data>
<session name="NumTel"/>
</data>
</substitute>
</configuration>

The VXML GW thrown error.connection.baddestination event to CVP application, and there is this message error on the VXML GateWay : ContactingDest_SetupDone: Setup done status = CS_INVALID_Number



Could you please let me know how to fix this issue?

BR,
Maryam

[Chintan Gajjar]

must be something wrong with the gateway configuration as its not able to route the call. can you please post the number you are trying to transfer and also if possible please attach the gateway configs.

[maryamsharifi]

Hi Chintan,

Please find here the log and gateway config (Test call was made at 15/04/2016 at 4:35PM)

The number we are trying to transfer is 0617834505.

Thanks for your help

Maryam

[Chintan Gajjar]

I am looking at the logs and trying to understand the call flow.

where 0617834505 should get routed? to PSTN via pots dialpeer or to CUSP via SIP trunk?

[Chintan Gajjar]

Why do i see the call coming from SIP Dialer? whats the call flow here?

[maryamsharifi]

Call flow is:  mobile phone à Ingress gateway (E1) à CUSP à CVP (Call studio script with a bridge transfer) à VXML gateway. And then it’s not going further with a VXML error.

Before we had on gateway for incoming/outgoing calls (E1) AND VXML. Now we have 1 gateway for incoming/outgoing calls and 1 new gateway dedicated for VXML (for a better load re-partition)

Before this change the same call flow works well but since this change no more transfer from CVP.

[Chintan Gajjar]

I don't see the Dial-peer on Your VXML gateway for the pattern 0617 for sending calls to voice gateway having E1 connection.

You have to configure it.

Regards

Chintan

[maryamsharifi]

Hi Chintan,

0617xxxxxx is the external phone number ( could be client or any external phone number), so are you sure that we should  set the Dial-peer on VXMLGW for each external phone number? if yes the VXML GW configuration should changed systematically ................

BR,

Maryam

[Chintan Gajjar · Accepted solution]

yes, otherwise how would VXML gateway know that where to route the PSTN calls?

the transfer element uses Voice Browser on VXML gateway to initiate outbound calls, so your VXML gateway should be aware of routing.

you can put cath all dial-peer (Destination Pattern .T) on VXML gateway that route the PSTN calls to ingress gateway and ingress gateway can use PRIs to send them out to PSTN.

[maryamsharifi]

Hi Chintan,

Thanks a lot for your help, after adding the dial-peer the transfer works well.

Regards,

Maryam