Not sure it makes any difference in this situation, but I never use codec pass-through on my configuration. I've never had any issues.
On Wed, Mar 6, 2013 at 12:32 PM, <michael.se...@compucom.com> wrote: > --MJ > > Your problem is a misconfigured location somewhere in CUCM. > > Your configuration on the gateways is correct to allow 4 calls using RSVP > based CAC. In my experience the issue your running into is not going to be > an issue with the configuration on your gateways (use show SCCP on gateways > to verify media resource registration), but a misconfigured location in > CUCM of an assignment of a location either on phone, gateway or device > pool. Not only are your calls not invoking CAC/AAR but they are NOT > rerouting which points to your Route Patterns/Route List configuration. > You might also verify the mask on your phones regarding AAR kicking in as > well as applying the AAR calling search space on the gateways and the > Device level of the phone. You also need to apply the AAR group to the > gateway and Phone device level. On the live level you must also set the > AAR group. > > Michael Sears > CCIE (V) 38404 > > > 2. RSVP a big problem (sanity insanity) > > > ---------------------------------------------------------------------- > > Message: 2 > Date: Wed, 6 Mar 2013 21:49:54 +0530 > From: sanity insanity <networksanitytoinsan...@gmail.com> > To: ccie_voice@onlinestudylist.com > Subject: [OSL | CCIE_Voice] RSVP a big problem > Message-ID: > < > cag4zmyw5dpqbxmgrcj3finope+pnur8zbjepv26cywgqyfh...@mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > hi Guys, > > I have to Configure IP Phones and gateways in such as way that all calls > within same site should use G711 Codec. Also, all calls between the sites > to remote IP phones and gateways should use G729 Codec. > RSVP Call Admission Control (CAC) between HQ and branch site based on > bandwidth limitations. There can be 4 concurrent calls. G711 CODEC to be > used for multi-directional audio. > > Steps:- > > 1) I set the location Bw between my headquater and branch as Mandatory. > > 2) I also have the MTP registered and added to the correct MRG > MRGLs > > 3) The following is a snip of my config on headquarter... > > > dspfarm profile 1 mtp > no codec g711u > codec g729r8 > codec pass?through > rsvp > maximum sessions software 4 > associate application SCCP > ! > interface Serial0/0/0.2 point?to?point > ip rsvp bandwidth 112 # 4 call > > > similarly on branch site... > > > dspfarm profile 1 mtp > no codec g711u > codec g729r8 > codec pass?through > rsvp > maximum sessions software 4 > associate application SCCP > ! > interface Serial0/0/0.2 point?to?point > ip rsvp bandwidth 112 # 4 call > > > > Questions: > ================== > > 1) With the above config I notice that when I make a call from headquarter > site 2XXX to branch site 4XXX . The message on the phone is "Not enough > Bandwidth" and the call disconnects. > What is the exact problem? > > 2) Is my config above correct? > > > -MJ > > _______________________________________________ > 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
