Yep, my worry is that an evil SS7 link will dump the RDNIS and then the call will hit the opening greeting...
Jonathan On Thu, Apr 17, 2008 at 9:16 AM, Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) <[EMAIL PROTECTED]> wrote: > Thx Jonthan. According to Vik's reply earlier on this, it looks like > AAR/UCM builds the RDNIS correctly, and passes it on back to Unity. The > *only* thing is of course that the ISDN cloud, or Telco cloud in general > needs to pass this info too. If the telco cloud doesn't relay this RDNIS > (example: no ISDN), I'm not aware of another solution for this... For > SRST there's the VM-integration stuff, but that doesn't work in this > case. But I believe that was also what you were referring to right > Jonathan? > > > -----Original Message----- > From: Jonathan Charles [mailto:[EMAIL PROTECTED] > > Sent: Thursday, April 17, 2008 4:06 PM > To: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) > > > Cc: Gregory Jost (grjost); CCIE Voice > Subject: Re: [OSL | CCIE_Voice] Dial Plan design question, AAR with > vmail... > > Well, what is actually happening? > > Call comes in on BR1 router, tries to ring a phone (on the CCM > cluster) at BR1, RNA, gets forwarded to VM... > > OK. > > Who is calling VM? the Phone at BR1 that CFNA to VM? Or the gateway? > > From an IP perspective, the call leg is terminated and initiated by the > GW, not the phone at BR1... So, we will then have RDNIS back out the GW > to HQ and to Unity... > > My question is, will this RDNIS get passed to Unity when the call is > redirected and will it go to the right mailbox. > > From Unity's perspective, the original called party is the Unity pilot, > as the GW is initiating that call leg... How do we get that call into > the right mailbox???? > > > > Jonathan > > On Thu, Apr 17, 2008 at 8:53 AM, Juan Lopez Hernandez -X (jlopezhe - IBM > - INS at Cisco) <[EMAIL PROTECTED]> wrote: > > > > Ah ! Thanks Greg and Jonathan. > > I didn't consider the case when a PSTN caller called BR1 phone, > > forcing a hairpin on BR1 GW in case no BW left between the BR1 GW and > > > the HQ's VM. > > > > I suppose when BR1 phone redirects to Unity, signaling (ECS) sets up > > a new call between calling (BR1 GW) party and called party (Unity) > right? > > Just to make sure it's correct what I say above: 'no BW left between > > the > > BR1 GW and the HQ's VM' - or: CAC between calling (BR1 GW) and called > > > (Unity pilot) is considered, not CAC BR1phone - Unity pilot Cheers, > > Juan > > > > > > -----Original Message----- > > From: Gregory Jost (grjost) > > Sent: Thursday, April 17, 2008 3:18 PM > > To: Juan Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco); Jonathan > > > Charles; [EMAIL PROTECTED] > > Cc: CCIE Voice > > > > > > Subject: RE: [OSL | CCIE_Voice] Dial Plan design question, AAR with > > vmail... > > > > The gateway is calling device (on the VoIP network). In the case of > > a redirected call, the called party becomes the redirecting party, so > > > its AAR Group/CSS is not used. When Location bandwidth is exhausted, > > > the calling device needs to know the prefix (AAR Group) and > > appropriate gateway (AAR CSS) in order to re-route the call over > > PSTN. In this case, the gateway will hairpin back to PSTN. > > > > > > Greg Jost > > Network Consulting Engineer > > Unified Communications Practice > > Cisco Systems, Inc. > > 214-274-1922 > > > > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Juan > > Lopez Hernandez -X (jlopezhe - IBM - INS at Cisco) > > Sent: Thursday, April 17, 2008 6:51 AM > > To: Jonathan Charles; [EMAIL PROTECTED] > > Cc: CCIE Voice > > Subject: Re: [OSL | CCIE_Voice] Dial Plan design question, AAR with > > vmail... > > > > I'm missing the point why we need the AAR CSS and AAR group on the > > remote GW for redirected calls? > > > > Cheers, > > Juan > > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan > > > Charles > > Sent: Wednesday, April 16, 2008 10:43 PM > > To: [EMAIL PROTECTED] > > Cc: CCIE Voice > > Subject: Re: [OSL | CCIE_Voice] Dial Plan design question, AAR with > > vmail... > > > > That makes sense... > > > > Now, what about when a GK denies the call to insufficient bandwidth? > > > > The only thing I can think of is to put an H.323 gateway second the > > route-list for the GK-controlled trunk (from CCM) and turn off the > > stop hunting on unallocated/busy/etc... > > > > > > > > Jonathan > > > > On Wed, Apr 16, 2008 at 3:12 PM, Vik Malhi <[EMAIL PROTECTED]> > wrote: > > > AAR cannot be "down" and AAR can't deny a call. Your entire > > > CallManager cluster is either unavailable (SRST is the solution) or > > > it is available (AAR being the solution when the WAN is saturated). > > > If CallManager is active and alive then AAR cannot fail unless off > > > > course it is misconfigured or you are out of B-channels on your > > PSTN connection. > > > > > > Assuming the CallManager is operational and the remote sites > > phones > and gateway are still registered then AAR is the solution > > for when > Locations CAC blocks the call. So on the remote phon you > > hit the > "Messages" button and CallManager determines there is no > > Locations > bandwidth available. The External Number Mask and AAR > > Group needs to > be configured on the Hunt Pilot for Voicemail. The > > AAR CSS and AAR > Group needs to be configured on the calling > > (remote) phone. Also in > the case of a Call Forward from the remote > > > phone, the remote gateway > needs an AAR CSS and AAR Group > (+Redirecting # outbound). > > > > > > Assuming the CallManager is not available (WAN outage) then SRST > > is > your only option. Nothing on CallManager works including AAR. > > > > > > > > > > > > Vik Malhi - CCIE #13890 > > > Senior Technical Instructor - IPexpert, Inc. > > > > > > Telephone: +1.810.326.1444 > > > Fax: +1.810.454.0130 > > > Mailto: [EMAIL PROTECTED] > > > > > > Join our free online support and peer group communities: > > > http://www.IPexpert.com/communities > > > > > > IPexpert - The Global Leader in Self-Study, Classroom-Based, > > > Video-On-Demand and Audio Certification Training Tools for the Cisco > > > > CCIE R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE > > > Voice Lab and CCIE Storage Lab Certifications. > > > > > > > > > -----Original Message----- > > > From: Jonathan Charles [mailto:[EMAIL PROTECTED] > > > Sent: > > Wednesday, April 16, 2008 1:01 PM > To: [EMAIL PROTECTED] > Cc: > > Onur Tufekci; CCIE Voice > Subject: Re: [OSL | CCIE_Voice] Dial Plan > > > design question, AAR with vmail... > > > > > > what I meant was SRST config to reroute the call if AAR is down... > > > > > > So, if I am at the remote site, AAR is denying calls and I hit the > > > > messages button, what are my options? > > > > > > > > > > > > Jonathan > > > > > > On Wed, Apr 16, 2008 at 2:57 PM, Vik Malhi <[EMAIL PROTECTED]> > > wrote: > > > > You can configure both- but only one of AAR and SRST will be > > active > > > > > at > any one point. When the WAN is saturated signaling still > > > traverses > the WAN. When there is a WAN outage then you lose > > > signaling to the > remote sites and SRST kicks in. From your > > > original question you > indicate that SRST might be a solution for > > AAR which it isn't. > > > > > > > > > > > > > > > > Vik Malhi - CCIE #13890 > > > > Senior Technical Instructor - IPexpert, Inc. > > > > > > > > Telephone: +1.810.326.1444 > > > > Fax: +1.810.454.0130 > > > > Mailto: [EMAIL PROTECTED] > > > > > > > > Join our free online support and peer group communities: > > > > http://www.IPexpert.com/communities > > > > > > > > IPexpert - The Global Leader in Self-Study, Classroom-Based, > > > > > Video-On-Demand and Audio Certification Training Tools for the > > Cisco > > > > > > CCIE R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , > > CCIE > > Voice Lab and CCIE Storage Lab Certifications. > > > > > > > > > > > > > > > > > > > > -----Original Message----- > > > > From: Jonathan Charles [mailto:[EMAIL PROTECTED] > Sent: > > > Wednesday, April 16, 2008 12:41 PM > To: [EMAIL PROTECTED] > > Cc: > > > > > Onur Tufekci; CCIE Voice > Subject: Re: [OSL | CCIE_Voice] Dial > > Plan > > > > > design question, AAR with vmail... > > > > > > > > Well the goal is two-fold. > > > > > > > > First, if the WAN is saturated, someone at HQ, should be able > > to > call > BR2 via the PSTN (using AAR), and a CFNA/CFB should > > route > back to > vmail and the correct box > > In the event of > > a WAN > outage, the phones in SRST should also CFNA/CFB > to > > voicemail and > the correct box > > > > > Jonathan > > On > > Wed, Apr 16, 2008 > at 2:23 PM, Vik Malhi <[EMAIL PROTECTED]> > wrote: > > > > > > > > > > > > > > > Good point- AAR requires you to configure an AAR CSS, AAR > > Group > on > the > remote site gateway in addition to checking the > > > > Redirecting > Number > Outbound checkbox (if this is MGCP then no > > > > mgcp/mgcp on the > IOS). You > need to check the Redirecting > > Number > Inbound checkbox on > the HQ > gateway. You also need an > > external > number mask and aar group > on the > hunt pilot. It > > should work a > treat having done this. The > Redirecting > Number > > is good since CCM > > > > > builds this in the case of > AAR. Only SRST > needs a workaround > > > > solution to get the caller to hear subscriber greeting. > > > > > > > > > > The SRST solution was outlined in a previous email. > > > > > > > > > > Don't overlap the two questions/solutions since they are > > > mutually > exclusive. > > > > > > > > > > > > > > > > > > > > Vik Malhi - CCIE #13890 > > > > > Senior Technical Instructor - IPexpert, Inc. > > > > > > > > > > Telephone: +1.810.326.1444 > > > > > Fax: +1.810.454.0130 > > > > > Mailto: [EMAIL PROTECTED] > > > > > > > > > > Join our free online support and peer group communities: > > > > > http://www.IPexpert.com/communities > > > > > > > > > > IPexpert - The Global Leader in Self-Study, Classroom-Based, > > > > > > > > > > Video-On-Demand and Audio Certification Training Tools for the > > Cisco > > > > > > > CCIE R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , > > CCIE > > > > > > > Voice Lab and CCIE Storage Lab Certifications. > > > > > > > > > > > > > > > ________________________________ > > From: > > > [EMAIL PROTECTED] > > > > > [mailto:[EMAIL PROTECTED] On Behalf Of > > > > Onur > > Tufekci > Sent: Wednesday, April 16, 2008 12:12 PM > > > > To: > > > > Jonathan Charles > Cc: CCIE Voice > Subject: Re: [OSL | > > > CCIE_Voice] > Dial Plan design question, AAR with vmail... > > > > > > > > > > > > > > > > > > > > Are you trying to configure only AAR or only SRST? > > > > > > > > > > > > > > > On Wed, Apr 16, 2008 at 2:03 PM, Jonathan Charles > > > > <[EMAIL PROTECTED]> > wrote: > > > > > > > > > > > OK, so AAR kicks in and I forward the call to the PSTN, the > > > > user > I > > was dialing does not answer and the call should > > forward to vmail... > > > > > > how would we make this work? voicemail under srst config? > > > > > > > > > > > > > > > > > > > > > > > > Jonathan > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >
