It may not be called SRST, but could your friend be referencing to the 'ip
source-address x.x.x.x secondary' command?
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_i1ht.html#wp1012400
**
(Optional) Second Cisco Unified CME router with which phones can register if
the
FR = 4 bytes
FRF.12 = 8 bytes
Agreed.
For MLPoFR (w/ or w/out LFI - but in our case we would only be using MLP for
LFI) I've been using 4B (FR) + 13B (MLP). Also, in all the IPExpert solution
guides for Volume 3 atleast they've been using 4B+13B for MLPoFR
On Wed, Jan 14, 2009 at 1:50 PM, Ryan
I understand that when configuring LLQ w/ FR, TS is required. If we
configured FRF.12, TS is also required.
Several questions;
1. If we configure MLP alone, is TS required?
I was under the assumption that it's not required. Going by Volume 3 L5 Q43,
the solutions doesn't have TS configured when
>From the CME router, can you collect the following debugs and send it as a
txt attachment?
debug voip ccapi inout
debug ras
debug h225 asn1
debug h245 asn1
Configure 'service sequence-numbers' as well.
On Sun, Jan 11, 2009 at 2:53 PM, Ryan Trauernicht
wrote:
> The TCS was unchecked already. M
AFAIK, the GK will load balance (random select) b/w the endpoints registered
to the zone that you are trying to hopoff to. I don't believe there's a way
to set a preference among the endpoints registered to the hopoff zone.
If you don't want CM to register w/ a TP, can you configure a TP on the GK
In the dial plan section of the solutions for Volume 3 Lab 7 the solutions
mention - "Make sure that the 'Provide Outside Dialtone' checkbox is NOT
marked on any of the Route Patterns,..." - pg 86 right below the dial plan
table.
I'm guessing that they mean the reverse?
For DSP calculations for voice termination-
PVDM-12's -
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a00800b65d6.shtml
Each PVDM-12 contains 3 TI 549 DSPs.
-
Runs up to twelve voice calls using a medium complexity CODEC (G.711,
G.729a/b, G.726, fax).
-
It should work w/ SIP. The only snag you can run into is if you configure
the dtmf-relay types to be different for the outbound/inbound SIP dial-peer.
Can you try changing dial-peer 3500 to the following and test?
dial-peer voice 3500 voip
service aa
destination-pattern 3500
session target ipv4
Thanks Ryan!
On Tue, Jan 6, 2009 at 10:45 PM, Ryan Trauernicht
wrote:
> It would be post compression.
>
> The priority queue is based on "output" assuming you are applying it on the
> output of the interface.
>
> Thanks,
> Ryan Trauernicht
>
>
> On Tue,
If we apply cRTP on the FR interface using "frame-relay ip rtp
header-compression", would our priority-map bandwidth values for rtp be
based on pre-compression or post-compression?
interface Serial0/0.1 point-to-point
ip address 1.1.1.1 255.255.255.0
frame-relay class fr-mapclass
frame-relay in
10 matches
Mail list logo
