[cisco-voip] SRST with Call manager 5.0

[zohaib shabir]

Thank you for your reply Erick
i will definitely go for H.323 for signaling on WAN but client requires SIP
phones both cisco and non-cisco at central site and remote site. Will the
sip clients work with H.323 on WAN links for signaling and will they work
normally on failover as SCCP clients work.


On 4/4/07, Erick Bergquist <erickbe at yahoo.com> wrote:
>
> 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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-- 
Regards,
Zohaib Shabir
Network Engineer(Voice and Presales)
DWP Group, TECH Division
Ph:+92-302-8232689
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