Hi Alex , Did you enabled automated alternate routing in service parameter ?? On Aug 23, 2013 8:41 PM, <ccie_voice-requ...@onlinestudylist.com> wrote:
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Re: AAR Not Attempting to ReRoute (Martin Sloan) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 22 Aug 2013 17:00:58 -0400 > From: Alex Pishko <alexpis...@gmail.com> > To: ccie_voice@onlinestudylist.com > Subject: [OSL | CCIE_Voice] AAR Not Attempting to ReRoute > Message-ID: > <CADL2L58Vx=t71ck5Y2hcaTr8+zWAatnwanr8= > jpgj1t55yf...@mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > All, > > Hopefully someone can give me an idea on this one. Working on my own > equipment but following the labs and am running into an issue with AAR. > > I'm sending a call from HQ to BR 1 using 4 digit dial 5002 ---> 1002. When > I attempt to make the call I receive the message Not enough bandwidth, > however I never see AAR actually get invoked. In my setup as a quick way > to simulate congestion I set the bandwidth to 23 Kbps between Hub non (hq) > and BR1. > > When I place the call from HQ as I said I get the banner of not enough > bandwidth but I never see the call actually hit the HQ gateway. I've run > debug voip dialpeer, q931 as well voice ccapi inout and neither shows any > sort of traffic hitting the gateway. > > In testing I can successfully dial into the HQ GW, I can dial emergency > services from the HQ phone, just doesn't seem like it's ever invoking AAR. > > I also checked that the external number mask is correctly defined on the > 1002 extension. > > AAR CSS is assigned to the HQ phone, AAR group is assigned to the HQ line. > AAR group is prefixing 91 and there is a RP assigned to a partition that > falls within the AAR CSS that is for 91617XXXXXXX that has a RL pointed to > the HQ GW. > > I did see earlier in the lab that they recommend using 7962 phones, however > I don't have any avaialble at the moment, so just wanted to make sure that > this might not be it. > > Any help would be much appreciated. > > Thank you > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > </archives/ccie_voice/attachments/20130822/3384f9de/attachment-0001.html> > > ------------------------------ > > Message: 2 > Date: Thu, 22 Aug 2013 23:35:14 -0400 > From: Justin Carney <justin.s.car...@gmail.com> > To: Karen Johnson <karen.johnson...@yahoo.ca> > Cc: "ccie_voice@onlinestudylist.com, > \(ccie_voice@onlinestudylist.com\)" < > ccie_voice@onlinestudylist.com> > Subject: Re: [OSL | CCIE_Voice] srst mode all -- remove mode all > Message-ID: > < > caex4d8xr-_u1kar0gza2akbucwuycwnkwmx3+qa8unxtrof...@mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > The command "srst mode auto-provision all" tells the router that "when you > learn info from a phone in srst mode (via SNAP) keep that info in running > config." If you desire to keep all that info in running config you also > need to keep the command in when saving the config. > > Althought I haven't tested it, I would expect that removing this auto > provision all (or dn) would also remove the dynamically learned info from > the running config but any command you manually typed (such as a hunt list) > would stay. If you let srst learn the ephone dn dynamically, then you > manually add a description (as in your original question) I would not > expect than ephone dn or description to show in running config after > removing the auto provision all. > > The link below discusses the various srst options and gives many cli > examples of features you can deploy with cme as srst and using auto > provision all. I would suggest revieiwng these config samples to help > prepare for the lab. > > > http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmesrst.html > > I have heard the same advice for srst in the lab from several people: > configure srst, test it once and exit, then save and reload. Do not test > srst again, trust your config if it worked the first time. The reason for > this advice is that the older router code has some bugs and if you try to > enter/exit srst multiple times the first will work but it may fail on > subsequent tries. > On Aug 22, 2013 1:28 PM, "Karen Johnson" <karen.johnson...@yahoo.ca> > wrote: > > > hi Justion, > > > > then I after modify and before save, i also need to remove "srst auto > mode > > prov all" under telephony service, right ? > > > > K > > > > *From:* Justin Carney <justin.s.car...@gmail.com> > > *To:* Karen Johnson <karen.johnson...@yahoo.ca> > > *Cc:* "ccie_voice@onlinestudylist.com, (ccie_voice@onlinestudylist.com)" > < > > ccie_voice@onlinestudylist.com> > > *Sent:* Wednesday, August 21, 2013 8:37:11 PM > > *Subject:* Re: [OSL | CCIE_Voice] srst mode all > > > > Short answer - yes you need to save the config. > > Long answer - if you want to ensure the ephones, ephone-dns, and > > descriptions are all "shown in the running config" then you should save > the > > running config to startup config at a point when all the settings you > want > > (ie, description) are visible with "show run." The two good ways to do > > this are to manually type the config before entering srst and save, or > > trigger srst then modify config as needed EXIT srst and save. > > It is typically recommended to NOT save the running config while srst is > > active (if using mgcp) because the mgcp-fallback process will remove > > automatically remove the serial interface subcommand "isdn bind-l3 ccm" > > when srst is triggered allow the default application (h323) to contol the > > serial interface. When cucm is reachable again this same process will > > restore the bind command to allow l3 backhaul of the serial port to cucm. > > So, if you save your config while in srst and then you (or the grading > > script) reload your router the bind command will be missing and the > serial > > port layer 3 will not come up until you manually put it back. > > I just make it a habit to not save any configs while in srst mode, but if > > you are using h323 there may not be any issues - someone else may be able > > to comment on that or you could try it as an experiment and let the group > > know how it goes. > > On Aug 21, 2013 9:43 PM, "Karen Johnson" <karen.johnson...@yahoo.ca> > > wrote: > > > > > > folks, > > > > when we have to use "srst auto mode provision all" and I need to change > > "Desctiption " of ephone-dn > > After come back to Normal mode. do we need to save the config or just > > leave it? > > > > K > > > > _______________________________________________ > > For more information regarding industry leading CCIE Lab training, please > > visit http://www.ipexpert.com/ > > > > Are you a CCNP or CCIE and looking for a job? Check out > > http://www.platinumplacement.com/ > > > > > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > </archives/ccie_voice/attachments/20130822/8e4f52f1/attachment-0001.html> > > ------------------------------ > > Message: 3 > Date: Thu, 22 Aug 2013 23:43:33 -0400 > From: Justin Carney <justin.s.car...@gmail.com> > To: Karen Johnson <karen.johnson...@yahoo.ca> > Cc: "ccie_voice@onlinestudylist.com, > \(ccie_voice@onlinestudylist.com\)" < > ccie_voice@onlinestudylist.com> > Subject: Re: [OSL | CCIE_Voice] MOH multicast - need Multicast ? > Message-ID: > < > caex4d8ztoqcc7i6svfxt72br2aarewug7tvu1zhc8xk5s0g...@mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > You don't need those commands on site b in the following scenarios: > 1-you are using unicast moh > 2-you are using moh from flash at site b > > You *should* need those commands to achieve multicast moh from site a cucm > over wan to site b phones. > On Aug 22, 2013 2:46 PM, "Karen Johnson" <karen.johnson...@yahoo.ca> > wrote: > > > > > all, > > > > i am just curious, my MOH SB for multicast always work without "ip > > multicast-routing" and "ip pim dense-mode" > > > > do we really need enable MCAST for moh to work ? And why without > Mcats > > it also works ? > > > > K > > > > _______________________________________________ > > 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 > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > </archives/ccie_voice/attachments/20130822/bc1a8fd6/attachment-0001.html> > > ------------------------------ > > Message: 4 > Date: Fri, 23 Aug 2013 11:07:44 -0400 > From: Martin Sloan <martinsloa...@gmail.com> > To: Alex Pishko <alexpis...@gmail.com> > Cc: "ccie_voice@onlinestudylist.com" <ccie_voice@onlinestudylist.com> > Subject: Re: [OSL | CCIE_Voice] AAR Not Attempting to ReRoute > Message-ID: > < > camvpjhjwmym6hfuizwqhzzkscb8rrgkgtkzroncqvwcys_w...@mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi Alex, > > Have you enabled AAR in CCM service parameters? Not Enough Bandwidth > indicates that it's not enabled. You should receive something like > 'Network Congestion - Rerouting' when AAR is invoked. > > Marty > > > On Thu, Aug 22, 2013 at 5:00 PM, Alex Pishko <alexpis...@gmail.com> wrote: > > > All, > > > > Hopefully someone can give me an idea on this one. Working on my own > > equipment but following the labs and am running into an issue with AAR. > > > > I'm sending a call from HQ to BR 1 using 4 digit dial 5002 ---> 1002. > > When I attempt to make the call I receive the message Not enough > bandwidth, > > however I never see AAR actually get invoked. In my setup as a quick way > > to simulate congestion I set the bandwidth to 23 Kbps between Hub non > (hq) > > and BR1. > > > > When I place the call from HQ as I said I get the banner of not enough > > bandwidth but I never see the call actually hit the HQ gateway. I've run > > debug voip dialpeer, q931 as well voice ccapi inout and neither shows any > > sort of traffic hitting the gateway. > > > > In testing I can successfully dial into the HQ GW, I can dial emergency > > services from the HQ phone, just doesn't seem like it's ever invoking > AAR. > > > > I also checked that the external number mask is correctly defined on the > > 1002 extension. > > > > AAR CSS is assigned to the HQ phone, AAR group is assigned to the HQ > > line. AAR group is prefixing 91 and there is a RP assigned to a > partition > > that falls within the AAR CSS that is for 91617XXXXXXX that has a RL > > pointed to the HQ GW. > > > > I did see earlier in the lab that they recommend using 7962 phones, > > however I don't have any avaialble at the moment, so just wanted to make > > sure that this might not be it. > > > > Any help would be much appreciated. > > > > Thank you > > > > _______________________________________________ > > 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 > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: </archives/ccie_voice/attachments/20130823/030f3f88/attachment.html> > > ------------------------------ > > _______________________________________________ > CCIE_Voice mailing list > CCIE_Voice@onlinestudylist.com > http://onlinestudylist.com/mailman/listinfo/ccie_voice > > > End of CCIE_Voice Digest, Vol 90, Issue 39 > ****************************************** >
_______________________________________________ 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