this works fine in the ipexpert racks. call coming into BR1 via PSTN, rings BR1 phone, fwd to VM is blocked by CAC, AAR out of BR1 via PSTN to HQ PRI into voicemail with RDNIS end to end.
On Thu, Apr 17, 2008 at 10:19 AM, Jonathan Charles <[EMAIL PROTECTED]> wrote: > 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 > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >
