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<div dir="auto">Hy out there
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<div dir="auto">maybe this could be done with CURRI on an incomming route or translation pattern</div>
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<div dir="auto"><a href="https://devnetsupport.cisco.com/hc/en-us/sections/115002851247-Cisco-Unified-Routing-Rules-XML-Interface-CURRI-">https://devnetsupport.cisco.com/hc/en-us/sections/115002851247-Cisco-Unified-Routing-Rules-XML-Interface-CURRI-</a><br /></div>
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<div name="messageReplySection">On 19. Nov 2019, 16:15 +0100, Kent Roberts <kent@fredf.org>, wrote:<br />
<blockquote type="cite" class="spark_quote" style="margin: 5px 5px; padding-left: 10px; border-left: thin solid #1abc9c;">IPIVR. Well that’s been some time..... not sure on that one.... I don’t think that info ever makes it. I think your stuck with playing at the dialpeer level or the scripting. I’ll email if I think of anything <br />
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<div><br /></div>
Kent</div>
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<blockquote type="cite" class="spark_quote" style="margin: 5px 5px; padding-left: 10px; border-left: thin solid #e67e22;">On Nov 19, 2019, at 08:01, daniele visaggio <visaggio.daniele@gmail.com> wrote:<br />
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<div>Hi Kent,</div>
<div><br /></div>
<div>even though we are talking about a UCCE deployment, we are still stuck with IP IVR. This means no CVP for the time being.<br /></div>
<div><br /></div>
<div>The only way I can pass this kind of metadata to the IVR/ICM/UCCE world is through CTI/Jtapi, afaik.</div>
<div><br /></div>
<div>It is not clear to me the precise logic used by cucm to translate between sip and Jtapi.</div>
<div><br /></div>
<div>Any hints in this regard?</div>
<div><br /></div>
<div>Thanks!<br /></div>
</div>
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<div dir="ltr" class="gmail_attr">Il giorno mar 19 nov 2019 alle ore 15:23 Kent Roberts <<a href="mailto:kent@fredf.org">kent@fredf.org</a>> ha scritto:<br /></div>
<blockquote class="gmail_quote spark_quote" style="margin: 5px 5px; padding-left: 10px; border-left: thin solid #3498db;">
<div style="overflow-wrap: break-word;">Did something similar to this in the SBC at the dial-peer level with number translations, when UCCE first didn’t support improper ANI many moons ago...<br />
<div><br /></div>
<div>If you can grab the inbound call at the dial-peer level (or via the return carrier). And send it in to its own CUCM SIP config, then you can do anything you want with it.</div>
<div><br /></div>
<div>I believe your stuck replacing ANI, as CUCM may not forward all the sip headers…</div>
<div><br /></div>
<div>Have you tried to turn up the CVP SIP debugs, and see if the headers get passed?</div>
<div><br /></div>
<div><br />
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<div>On Nov 19, 2019, at 3:19 AM, daniele visaggio <<a href="mailto:visaggio.daniele@gmail.com" target="_blank">visaggio.daniele@gmail.com</a>> wrote:</div>
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<div>Thanks, Stephen.<br /></div>
<div><br /></div>
<div>Yes, I'm aware of lua scripting.<br /></div>
<div><br /></div>
<div>Having an sbc in front of the cucm, I already tried to alter the REFER message in some obvious ways but no luck so far.</div>
<div><br /></div>
<div>I tried also to transform the incoming REFER into a brand new INVITE (oracle sbc has this feature built-in). Sadly this breaks the routing, meaning the transfer totally fails.</div>
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<div>Before going on with other exotic manipulations, I would like to know in advance if what I want is even possible...it seems to me cucm is totally ignoring whatever I put in the REFER.<br /></div>
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<div>Best Regards<br /></div>
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<div dir="ltr" class="gmail_attr">Il giorno mar 19 nov 2019 alle ore 11:01 Stephen Welsh <<a href="mailto:stephen.welsh@unifiedfx.com" target="_blank">stephen.welsh@unifiedfx.com</a>> ha scritto:<br /></div>
<blockquote class="gmail_quote spark_quote" style="margin: 5px 5px; padding-left: 10px; border-left: thin solid #34495e;">
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<div>Hi Daniele,</div>
<div><br /></div>
Not my area, but have you looked at using LUA scripts to pass-thru/transform SIP headers on UCM:
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<div>
<div><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" target="_blank">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><br /></div>
<div>Thanks</div>
<div><br /></div>
<div>Stephen Welsh</div>
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<blockquote type="cite" class="spark_quote" style="margin: 5px 5px; padding-left: 10px; border-left: thin solid #2ecc71;">
<div>On 19 Nov 2019, at 09:38, daniele visaggio <<a href="mailto:visaggio.daniele@gmail.com" target="_blank">visaggio.daniele@gmail.com</a>> wrote:</div>
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<div dir="ltr">Good morning.<br />
<br />
Diagram:<br />
<br />
FINESSE --- UCCE --- CUCM --- SBC --- THIRD PARTY SIP SERVER<br />
<br />
<b>Scenario</b>:<br />
<br />
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 />
<br />
So we have:<br />
<br />
<b>Transferee</b>: it's the PSTN caller, i.e. the party ending up being transferred to the finesse agent<br />
<br />
<b>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 />
<br />
<b>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 />
<br />
In order to transfer the call, THIRD PARTY SIP SERVER sends a SIP REFER message to SBC/CUCM.<br />
<br />
From a routing perspective, the transfer works fine. The pstn caller can be transferred to a finesse agent.<br />
<br />
<b>GOAL</b>:<br />
<br />
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 />
<br />
There are business reasons why we need to do so.<br />
<br />
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 />
<br />
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 />
<br />
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 />
<br />
<div>Thanks,</div>
<div><br /></div>
<div>Daniele<br /></div>
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