[cisco-voip] Call Transfer: Inject an arbitrary calling id (ANI) to a JTapi Transfer target

Stephen Welsh stephen.welsh at unifiedfx.com
Tue Nov 19 05:01:07 EST 2019


Hi Daniele,

Not my area, but have you looked at using LUA scripts to pass-thru/transform SIP headers on UCM:

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/sip_tn/9_0_1/sip_t_n/5-sip_pass_thru.html

Thanks

Stephen Welsh

On 19 Nov 2019, at 09:38, daniele visaggio <visaggio.daniele at gmail.com<mailto:visaggio.daniele at gmail.com>> wrote:

Good morning.

Diagram:

FINESSE --- UCCE --- CUCM --- SBC --- THIRD PARTY SIP SERVER

Scenario:

CUCM receives a call from PSTN. A route pattern sends the call to THIRD PARTY SIP SERVER which, in turn, transfers the call back to UCCE IVR SCRIPT via SBC/CUCM.

So we have:

Transferee: it's the PSTN caller, i.e. the party ending up being transferred to the finesse agent

Transfer Target: technically it's a CTI route point on CUCM, which triggers a UCCE script placing the call on a queue. It is the new party being introduced to the Transferee. In the end it represents a finesse agent.

Transferor: THIRD PARTY SIP SERVER, i.e. the party initiating the transfer of the Transferee (PSTN caller) to the Transfer target (finesse agent)

In order to transfer the call, THIRD PARTY SIP SERVER sends a SIP REFER message to SBC/CUCM.

>From a routing perspective, the transfer works fine. The pstn caller can be transferred to a finesse agent.

GOAL:

we need to alter the calling id seen by UCCE and then by Finesse Agent. Actually, the calling id (ANI) seen by UCCE/Finesse is the original PSTN phone number.

There are business reasons why we need to do so.

The crucial point is that THIRD PARTY SIP SERVER sends back to cucm a custom sip header in the REFER message containing the phone number needed to be seen by UCCE/Finesse. This can be different from the original PSTN ANI (e.g. the pstn call is anonymous). This new ANI is dynamic and so it's not always the same.

I tried with many sip manipulations on the SBC. I placed the new ANI into the REFER FROM sip header, in the Remote-Party-id, the PAI header. Nothing worked so far.

Is there a way to set a new ani in this call transfer scenario? I need to find a way to "convince" cucm to pass the new ANI via Jtapi to UCCE/IVR/Finesse. Is this possible?

Thanks,

Daniele
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