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Home > Archive > Voice over IP Cisco > April 2007 > SRST with Call manager 5.0
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| Author |
SRST with Call manager 5.0
|
|
| zohaib shabir 2007-04-03, 1:12 pm |
| 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
| |
| Jonathan Charles 2007-04-03, 1:12 pm |
| Yes.
However, if you have translations or other call treatment on CCM, that will
not occur. You need to handle this with local trans patterns or num-exp...
Jonathan
On 4/3/07, zohaib shabir <zohaibshabir@gmail.com> wrote:
>
> 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
> ________________________________________
_______
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
| |
| Erick Bergquist 2007-04-03, 1:12 pm |
| 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 w
hen 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 ca
lls 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@gmail.com>
To: cisco-voip@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
________________________________________
_______
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
________________________________________
________________________________________
____
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| |
| zohaib shabir 2007-04-03, 1:12 pm |
| 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@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@gmail.com>
> To: cisco-voip@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 ________________________________________
_______
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> ------------------------------
> Never miss an email again!
> Yahoo! Toolbar<http://us.rd.yahoo.com/evt=49938/*h.../features/mail/>alerts you the instant new Mail arrives.Check it out.
>
--
Regards,
Zohaib Shabir
Network Engineer(Voice and Presales)
DWP Group, TECH Division
Ph:+92-302-8232689
| |
| zohaib shabir 2007-04-03, 1:12 pm |
| 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@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@yahoo.com> wrote:
>
>
>
> --
> 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
| |
| Ryan Ratliff 2007-04-03, 1:12 pm |
| 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@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@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@gmail.com>
To: cisco-voip@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
________________________________________
_______
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
Never miss an email again!
Yahoo! Toolbar alerts you the instant new Mail arrives. Check it out.
--
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@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
| |
| Erick Bergquist 2007-04-03, 7:11 pm |
| 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 s
etup 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@cisco.com>
To: zohaib shabir <zohaibshabir@gmail.com>
Cc: Erick Bergquist <erickbe@yahoo.com>; cisco-voip@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@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@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@gmail.com>
To: cisco-voip@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
________________________________________
_______
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
Never miss an email again!
Yahoo! Toolbar alerts you the instant new Mail arrives. Check it out.
--
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@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
________________________________________
________________________________________
____
Don't get soaked. Take a quick peek at the forecast
with the Yahoo! Search weather shortcut.
http://tools.search.yahoo.com/shortcuts/#loc_weather
| |
| zohaib shabir 2007-04-04, 1:11 am |
| 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@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@cisco.com>
> To: zohaib shabir <zohaibshabir@gmail.com>
> Cc: Erick Bergquist <erickbe@yahoo.com>; cisco-voip@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@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@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@gmail.com>
> To: cisco-voip@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
> ________________________________________
_______
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> Never miss an email again!
> Yahoo! Toolbar alerts you the instant new Mail arrives. Check it out.
>
>
>
> --
> 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@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
>
>
> ________________________________________
________________________________________
____
> Don't get soaked. Take a quick peek at the forecast
> with the Yahoo! Search weather shortcut.
> http://tools.search.yahoo.com/shortcuts/#loc_weather
>
--
Regards,
Zohaib Shabir
Network Engineer(Voice and Presales)
DWP Group, TECH Division
Ph:+92-302-8232689
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