Thanks, Stephen. Yes, I'm aware of lua scripting.
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. 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. 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. Best Regards Il giorno mar 19 nov 2019 alle ore 11:01 Stephen Welsh < stephen.we...@unifiedfx.com> ha scritto: > 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.dani...@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 > _______________________________________________ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > > >
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