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New Member

SIP Trunk Dial-Peer Configuration for Cisco CallManager Server Redundancy

Hello,

we would like to make a "SIP Trunk Dial-Peer Configuration for Cisco CallManager Server Redundancy".

That is, we would like to get in SIP what can be obtained with H.323, as in the document "H.323 Gateway Dial-Peer Configuration for Cisco CallManager Server Redundancy":

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a0080094852.shtml

Anyone could post a similar configuration, but for SIP?

TIA and regards.

6 REPLIES

SIP Trunk Dial-Peer Configuration for Cisco CallManager Server R

Make a second dial-peer to the CUCM, with the preference command to make redudancy.

Thanks

Hall of Fame Super Silver

SIP Trunk Dial-Peer Configuration for Cisco CallManager Server R

Same concept for SIP as H.323, if you want to consoldidate dial peers you can look into building static srv records on the GW and balance calls that way.

Chris

New Member

SIP Trunk Dial-Peer Configuration for Cisco CallManager Server R

Hello,

I just have created the second dial-peer, but it isn't enough.

I think that I need in SIP a configuration similar to this one (valid for H.323):

voice class h323 1

h225 timeout tcp establish 3

How can it be obtained in SIP?

Regards.

SIP Trunk Dial-Peer Configuration for Cisco CallManager Server R

Soory but you put the command preference under the dial-peer

Lower the preference, higher precedence like this way. If you are using Sip to the CUCM use the command session protocol sipv2.

Because per default the dial-peer uses H323

Thanks

New Member

SIP Trunk Dial-Peer Configuration for Cisco CallManager Server R

Hello,

"preference" and "session protocol sipv2" have been configured correctly on the 2 dial peers...

I repeat my question: how can I configure the timeout in SIP, so that (for example) 3 seconds, if with the first dial peers the call can't be routed to the first CUCM (because it's down, for example), the second dial peer is used to route the call to the second CUCM?

At the moment, this doesn't happen, or if it happens, it happens after a LOT of seconds.

Regards.

Hall of Fame Super Silver

SIP Trunk Dial-Peer Configuration for Cisco CallManager Server R

You define these under sip-ua, i.e.

sip-ua

retry invite 2

retry bye 2

retry cancel 2

retry options 1

timers expires 60000

as well as timers on CUCM via service parameters:

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_configuration_example09186a008082d76a.shtml

HTH, please rate all useful posts!

Chris

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