[OSL | CCIE_Voice] SIP Trunk with CUC and possible AAR
Hi All, A big thanks to Ash, Amit and Raees for replying...you guys gave me a lot of good information to think about. Before posting this issue I was about 90% of the way there...please see the scenario in my original email below. As you all probably know AAR is not supported with SIP trunks. Therefore in order to provide HA in the situation where you have WAN congestion, a route list with sip trunk as the first choice and PSTN gateway as the second choice must be used. However the problem occurs when trying to redirect to voicemail on NoAnswer or Busy. In this case if the CUC is to recognize the redirecting number you either have to forward four digits out the PSTN or at the HQ site you need to create a translation pattern and use the calling party mask to cut the number down to four digits. I hope this helps others in their journey. Cheers!! Inder. To: ccie_voice@onlinestudylist.com > Subject: [OSL | CCIE_Voice] SIP Trunk with CUC and possible AAR > Message-ID: > > > Content-Type: text/plain; charset="iso-8859-1" > > Hello All, > > I am working on a lab that requires to set up CUCM and CUC using SIP Trunk. > It then asks for calls that are sent to VM from BR1 to be redirected out > the > PSTN when there is WAN congestion. > > I have looked high and low but I can't find any reference where this can be > done with AAR...or am I totally missing something. If AAR is possible can > someone point me in the right direction? If it is not possible can someone > let me know how you might achieve this otherwise? > > I tried using a route list with the SIP trunk as the primary RG and the > PSTN > GW and the secondary RG. The issue is redirecting the caller, called and > redirect on no answer. We need the BR1 phone to be able to press the > message key and retrieve messages (this I was able to do with alternate > extensions) and also for callers to be redirected to voicemail for the > called party (this I was not able to do with the route list scenario). > > Thanks in advance for any help you can provide. > > Regards. Inder. > ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SIP Trunk with CUC and possible AAR
i guess you can take of the redirected number from the call and then do mask on the VM pilot /profile or interduce tranlation pattern in between to match the called and change the calling to 4 digits Ash On Thu, Oct 20, 2011 at 3:57 PM, Ashraf Ayyash wrote: > Hey Inder , > > Nice catch on this , the AAR is between 2 endpoint register to the > same ccm so you cannot use the same concept of redundancy in this > config scenario , you can achieve this by having 2 route in the route > group , so the sip trunk to the CUC and also GW (of course manipulate > the calling and the called to be fit to the PSTN world) and change the > service parameter of stop routing when no bandwidth to false , and so > you will accomplish the redundancy in this config scenario > > Ash > > On Thu, Oct 20, 2011 at 10:48 AM, Inder Singh wrote: >> Hello All, >> >> I am working on a lab that requires to set up CUCM and CUC using SIP Trunk. >> It then asks for calls that are sent to VM from BR1 to be redirected out the >> PSTN when there is WAN congestion. >> >> I have looked high and low but I can't find any reference where this can be >> done with AAR...or am I totally missing something. If AAR is possible can >> someone point me in the right direction? If it is not possible can someone >> let me know how you might achieve this otherwise? >> >> I tried using a route list with the SIP trunk as the primary RG and the PSTN >> GW and the secondary RG. The issue is redirecting the caller, called and >> redirect on no answer. We need the BR1 phone to be able to press the >> message key and retrieve messages (this I was able to do with alternate >> extensions) and also for callers to be redirected to voicemail for the >> called party (this I was not able to do with the route list scenario). >> >> Thanks in advance for any help you can provide. >> >> Regards. Inder. >> >> ___ >> For more information regarding industry leading CCIE Lab training, please >> visit www.ipexpert.com >> >> Are you a CCNP or CCIE and looking for a job? Check out >> www.PlatinumPlacement.com >> > ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SIP Trunk with CUC and possible AAR
Hey Inder , Nice catch on this , the AAR is between 2 endpoint register to the same ccm so you cannot use the same concept of redundancy in this config scenario , you can achieve this by having 2 route in the route group , so the sip trunk to the CUC and also GW (of course manipulate the calling and the called to be fit to the PSTN world) and change the service parameter of stop routing when no bandwidth to false , and so you will accomplish the redundancy in this config scenario Ash On Thu, Oct 20, 2011 at 10:48 AM, Inder Singh wrote: > Hello All, > > I am working on a lab that requires to set up CUCM and CUC using SIP Trunk. > It then asks for calls that are sent to VM from BR1 to be redirected out the > PSTN when there is WAN congestion. > > I have looked high and low but I can't find any reference where this can be > done with AAR...or am I totally missing something. If AAR is possible can > someone point me in the right direction? If it is not possible can someone > let me know how you might achieve this otherwise? > > I tried using a route list with the SIP trunk as the primary RG and the PSTN > GW and the secondary RG. The issue is redirecting the caller, called and > redirect on no answer. We need the BR1 phone to be able to press the > message key and retrieve messages (this I was able to do with alternate > extensions) and also for callers to be redirected to voicemail for the > called party (this I was not able to do with the route list scenario). > > Thanks in advance for any help you can provide. > > Regards. Inder. > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > Are you a CCNP or CCIE and looking for a job? Check out > www.PlatinumPlacement.com > ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] SIP Trunk with CUC and possible AAR
Hello All, I am working on a lab that requires to set up CUCM and CUC using SIP Trunk. It then asks for calls that are sent to VM from BR1 to be redirected out the PSTN when there is WAN congestion. I have looked high and low but I can't find any reference where this can be done with AAR...or am I totally missing something. If AAR is possible can someone point me in the right direction? If it is not possible can someone let me know how you might achieve this otherwise? I tried using a route list with the SIP trunk as the primary RG and the PSTN GW and the secondary RG. The issue is redirecting the caller, called and redirect on no answer. We need the BR1 phone to be able to press the message key and retrieve messages (this I was able to do with alternate extensions) and also for callers to be redirected to voicemail for the called party (this I was not able to do with the route list scenario). Thanks in advance for any help you can provide. Regards. Inder. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
