UCCE11.5 Outbound Option- Transfer to IVR issue

Answered Question
Jul 10th, 2017
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I have configured the Outbound Option as per the guide but it is not working. 

A process status of Outbound Options Dialer is  good (ipcc-Dialer BADialer_SIP -A [CM-A] [CTI-A] [Ports C:4,R:4,B:0] [MR-A] [SIP-A]) 

Attached are CampaignManager and baDialer log.

Hope it helps.

Correct Answer by Chintan Gajjar about 2 weeks 1 day ago

I see SIP 403 received from the gateway when Dialer Sends out an invite.

Check if your gateway has dialed IP added in IP Address Trusted list.


13:48:33:164 dialer-baDialer Trace: (SipPort)	HandleInviteMsg(): initialRequest - SipReq:  INVITE [email protected] cid=d94ace7f-0a7b572e-b4272862-7604e121 tid=d42c55743544a535 cseq=INVITE contact=7800 / 1 from(tu). 
13:48:33:200 dialer-baDialer Trace: (RESIP) Failure:  error response: SipResp: 403 cid=d94ace7f-0a7b572e-b4272862-7604e121 tid=d42c55743544a535 cseq=INVITE / 1 from(wire) 
13:48:33:200 dialer-baDialer Trace: (RESIP) Transition UAC_Start -> InviteSession::Terminated 
13:48:33:200 dialer-baDialer Trace: (SipISH)	onFailure: SipResp: 403 cid=d94ace7f-0a7b572e-b4272862-7604e121 tid=d42c55743544a535 cseq=INVITE / 1 from(wire) 
13:48:33:200 dialer-baDialer Trace: (SipISH)	SIP port: [000], station: [7800], state: [SIP_LINESTATE_DIALING_NUMBER]. 
13:48:33:200 dialer-baDialer Trace: (SipISH)	Reason: Q.850;cause=21 
13:48:33:200 dialer-baDialer Trace: (SipISH)	onTerminated: 2SipResp: 403 cid=d94ace7f-0a7b572e-b4272862-7604e121 tid=d42c55743544a535 cseq=INVITE / 1 from(wire) 
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Correct Answer
Chintan Gajjar Mon, 07/10/2017 - 02:55
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I see SIP 403 received from the gateway when Dialer Sends out an invite.

Check if your gateway has dialed IP added in IP Address Trusted list.


13:48:33:164 dialer-baDialer Trace: (SipPort)	HandleInviteMsg(): initialRequest - SipReq:  INVITE [email protected] cid=d94ace7f-0a7b572e-b4272862-7604e121 tid=d42c55743544a535 cseq=INVITE contact=7800 / 1 from(tu). 
13:48:33:200 dialer-baDialer Trace: (RESIP) Failure:  error response: SipResp: 403 cid=d94ace7f-0a7b572e-b4272862-7604e121 tid=d42c55743544a535 cseq=INVITE / 1 from(wire) 
13:48:33:200 dialer-baDialer Trace: (RESIP) Transition UAC_Start -> InviteSession::Terminated 
13:48:33:200 dialer-baDialer Trace: (SipISH)	onFailure: SipResp: 403 cid=d94ace7f-0a7b572e-b4272862-7604e121 tid=d42c55743544a535 cseq=INVITE / 1 from(wire) 
13:48:33:200 dialer-baDialer Trace: (SipISH)	SIP port: [000], station: [7800], state: [SIP_LINESTATE_DIALING_NUMBER]. 
13:48:33:200 dialer-baDialer Trace: (SipISH)	Reason: Q.850;cause=21 
13:48:33:200 dialer-baDialer Trace: (SipISH)	onTerminated: 2SipResp: 403 cid=d94ace7f-0a7b572e-b4272862-7604e121 tid=d42c55743544a535 cseq=INVITE / 1 from(wire) 
lee yongboem Mon, 07/10/2017 - 19:40
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Thanks for your quick reply.

I'll check to VG configurations.


Chintan Gajjar Wed, 07/12/2017 - 20:53
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under your translation route configuration i see label is going to cucm routing client. why?

it should go to the MR routing client for dialer.

now dialer will get the label and has to connect it to cvp, for that you can define dial peer on voice gateway dialer connected to or on cups which should send it to cvp.


by the way you can also do this using send to vru node, but you have to setup the vru label for dialer flow correctly to reach to respective cvp servers.

Chintan is correct - you no longer need the Trans Route and can do it with the Send To VRU instead.

I never believed this would actually work until I tried it and was quite surprised to see it work. There is some magic going on behind the scenes with the correlation ID method of translation route - which is effectively what a Send to VRU node can do: get the call from one routing client to another routing client and keep the call context.

Just like a trans route can do.

And as everyone knows, getting that damn Translation Route Wizard to play nice is a lesson in frustration the first few (many?) times you try it.

And then suddenly, one day, it all clicks. It's an epiphany of sorts. ;-)

Regards,
Geoff

Chintan Gajjar Fri, 07/14/2017 - 02:54
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Its simple, in order to use Send To VRU instead of Translation route in Transfer to IVR campaign:

1) configure a Network VRU label under the Network VRU you are using for CVPs in your existing deployment.

Configure new label for Dialer Routing client, the length of the lebel should be the same as what you have for other network VRU labels.


2) when the Send to VRU is executed, the label you configured in above step prefixed by co-relation ID will be sent to Dialer Routing client and apparently to SIP dialer.

SIP Dialer will use this label and prepare a SIP REFER and send it out to the SIP endpoint its connected to it can be voice gateway and or Sip Proxy.


3) Configure Route on SIP PROXY or Dial-peer on a Gateway to Route the label (Label+corrid) to CVP call servers you wish the calls will be handled by.


Thats it, if things are setup correctly it should work like a charm.

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