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
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