<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=us-ascii">
</head>
<body style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class="">
<div class="">Hi Daniele,</div>
<div class=""><br class="">
</div>
Not my area, but have you looked at using LUA scripts to pass-thru/transform SIP headers on UCM:
<div class=""><br class="">
</div>
<div class="">
<div class=""><a href="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" class="">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</a></div>
<div class=""><br class="">
</div>
<div class="">Thanks</div>
<div class=""><br class="">
</div>
<div class="">Stephen Welsh</div>
<div><br class="">
<blockquote type="cite" class="">
<div class="">On 19 Nov 2019, at 09:38, daniele visaggio <<a href="mailto:visaggio.daniele@gmail.com" class="">visaggio.daniele@gmail.com</a>> wrote:</div>
<br class="Apple-interchange-newline">
<div class="">
<div dir="ltr" class="">Good morning.<br class="">
<br class="">
Diagram:<br class="">
<br class="">
FINESSE --- UCCE --- CUCM --- SBC --- THIRD PARTY SIP SERVER<br class="">
<br class="">
<b class="">Scenario</b>:<br class="">
<br class="">
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.<br class="">
<br class="">
So we have:<br class="">
<br class="">
<b class="">Transferee</b>: it's the PSTN caller, i.e. the party ending up being transferred to the finesse agent<br class="">
<br class="">
<b class="">Transfer Target</b>: 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.<br class="">
<br class="">
<b class="">Transferor</b>: THIRD PARTY SIP SERVER, i.e. the party initiating the transfer of the Transferee (PSTN caller) to the Transfer target (finesse agent)<br class="">
<br class="">
In order to transfer the call, THIRD PARTY SIP SERVER sends a SIP REFER message to SBC/CUCM.<br class="">
<br class="">
>From a routing perspective, the transfer works fine. The pstn caller can be transferred to a finesse agent.<br class="">
<br class="">
<b class="">GOAL</b>:<br class="">
<br class="">
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.<br class="">
<br class="">
There are business reasons why we need to do so. <br class="">
<br class="">
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.<br class="">
<br class="">
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.<br class="">
<br class="">
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?<br class="">
<br class="">
<div class="">Thanks,</div>
<div class=""><br class="">
</div>
<div class="">Daniele<br class="">
</div>
</div>
_______________________________________________<br class="">
cisco-voip mailing list<br class="">
<a href="mailto:cisco-voip@puck.nether.net" class="">cisco-voip@puck.nether.net</a><br class="">
https://puck.nether.net/mailman/listinfo/cisco-voip<br class="">
</div>
</blockquote>
</div>
<br class="">
</div>
</body>
</html>