[cisco-voip] SRST with Call manager 5.0

Tue Apr 3 15:59:38 EDT 2007

If the phones are to be registered to the central Call Manager for normal operation, then you'll always have the signalling traffic going over WAN between the phones and call manager. If the gateway is H.323 the H323 traffic will go across WAN when call setup and such happens. Once the call is established between the PRI and the IP phone local at the site, the RTP audio stream will be between the gateway and the IP Phone and not going across WAN unless you have resources configured in such a way the call needs to use MTP, etc across the WAN. To source MOH from router, you need to have MOH configured for multicast in CCM. 

However, any calls to anyone else at the other sites, then the RTP stream would go over the WAN. You could configure regions and do G729 and use locations to limit the bandwidth. You could also set up the partitions/CSSs in such a fashion so the people in this office can't call anyone else at other sites which will also prevent a RTP stream over WAN if you wanted to go to that extreme.
 would work out if bandwidth

----- Original Message ----
From: Ryan Ratliff <rratliff at cisco.com>
To: zohaib shabir <zohaibshabir at gmail.com>
Cc: Erick Bergquist <erickbe at yahoo.com>; cisco-voip at puck.nether.net
Sent: Tuesday, April 3, 2007 12:54:05 PM
Subject: Re: [cisco-voip] SRST with Call manager 5.0

RTP always goes between the endpoints involved in the call, unless a  
media resources is being invoked (MTP, CFB, etc).   In your case the  
remote site can be configured such that no RTP ever goes over the  
wan.   It will require DSP resources for transcoding and  
conferencing, MOH sourced from the flash of the router, and a  
dialplan such that any calls from the remote site phones to the  
central site go via the PSTN.

Newer versions of CME can work in SRST mode and this should allow you  
to use the tcl AA script on the router itself.  I don't believe CUE  
will work in SRST mode even with CME-SRST.

-Ryan

On Apr 3, 2007, at 1:47 PM, zohaib shabir wrote:

What actually my client want from me is that remote site Phones  
should be registered with Call Manager Cluster but calls coming on  
the PRI should be handled locally and no rtp stream should be  
created. Does SRST or unity express at remote site can act as IVR or  
autoattandant even when the phones are registered with Call manager  
Cluster.

On 4/3/07, zohaib shabir <zohaibshabir at gmail.com> wrote:
Thanyou Eirck and Jonathan

Does there exits anyway that calls can be handled locally not routed  
to CCM because the bandwidth here is pretty expensive and have to  
calculate the bandwidth of all incoming calls from remote site to  
central site. Tell me that it is possible that unity express can do  
it for me.

On 4/3/07, Erick Bergquist < erickbe at yahoo.com> wrote:
When the site isn't in SRST fallback mode, the inbound calls should  
be routed to Call Manager to be handled. If H.323, with dial-peers  
and if MGCP then the gateway and PRI endpoint will be registered to  
Call Manager and controlled by Call Manager itself when not in SRST  
mode and MGCP connection is active and registered.

If the gateway is MGCP, you will need to configure the ccm-manager  
for fallback-mgcp so the voice-ports on the router (PRI, etc)  
fallback to H323 mode when MGCP connection fails.  SRST and MGCP  
fallback are two seperate items.

In SRST Fallback (H323 also), you should have a pots dial-peer for  
the PRI with direct-inward-dial on it and then inbound calls will go  
to IP Phones registered to router in SRST mode. You may need to  
configure alias and/or translation rules if you have calls going to  
logical numbers (hunt groups, AAs, etc) so those numbers go to a real  
IP phone while router is in SRST mode.

HTH, Erick

----- Original Message ----
From: zohaib shabir < zohaibshabir at gmail.com>
To: cisco-voip at puck.nether.net
Sent: Tuesday, April 3, 2007 11:56:27 AM
Subject: [cisco-voip] SRST with Call manager 5.0

Dear All,

I have scenario where one side has call manager 5.0 cluster and the  
remote site has SRST enabled router with PRI line terminating on  
it.My question does SRST will be able handle all the PRI calls coming  
in locally or handled by call manager remotely.

-- 
Regards,
Zohaib Shabir
Network Engineer(Voice and Presales)
DWP Group, TECH Division
Ph:+92-302-8232689
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-- 
Regards,
Zohaib Shabir
Network Engineer(Voice and Presales)
DWP Group, TECH Division
Ph:+92-302-8232689

-- 
Regards,
Zohaib Shabir
Network Engineer(Voice and Presales)
DWP Group, TECH Division
Ph:+92-302-8232689
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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