SRST Fallback question

singhsaju · ‎04-19-2010

Hi

I have a SIP gateway router at one of our sites. This gateway has SIP trunk to SIP Provider for inbound and outbound calls. There are no PRI on this SIP gateway router.

The IP Phones at the site communicate with Call Manager over the WAN.There is H.323 between the Call Manager and this SIP gatway router.

Call manager fallback is configured on this SIP gateway router.

My question is when Call manager is unavailable how will the IP Phones at the site work? Would they be able to outbound and inbound calls after fall back to SIP gateway router? What all functions are avialable to IP Phone users when on fallback? What is not available?

Brandon Buffin · ‎04-19-2010

Your remote phones will work and be able to make and receive calls during fallback. See the following link for supported features in SRST.

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps2169/prod_qas0900aecd8028d113.html

Hope this helps.

Brandon

singhsaju · ‎04-19-2010

Hi Brandon ,

Thanks for the reply. Thanks for the link .

Following is the call manager fallback configuration:

call-manager-fallback
max-conferences 8 gain -6
transfer-system full-consult
limit-dn 7910 1
limit-dn 7935 1
limit-dn 7940 2
limit-dn 7960 4
timeouts interdigit 3
ip source-address 172.x.x.x port 2000
max-ephones 336
max-dn 960
dialplan-pattern 1 .... extension-length 5 extension-pattern 5....
transfer-pattern .....
default-destination 50800
alias 1 0 to 50800
call-forward busy 87006
call-forward noan 87007 timeout 20
date-format dd-mm-yy

The above configuration does not have any SIP related configuration.Is the configuration missing any commands for SIP? i forgot to mention that Ip phones are running SCCP firmware.

Not sure how will IP phones(after fallback) will make outbound/inbound calls through SIP provider .

What i understand that when IP phones will register with this gateway on fallback, the gateway will create a dial peer for each phone. And also DP for SIP provider is already configured on the gateway router. So using the dial peers the IP phones will be able to route calls in/out (without the CM) . Am i correct?

Thanks,

Saju

Brandon Buffin · ‎04-19-2010

Your phones will use the existing dial peers for outbound calls. From your config, looks like the telco is sending you 4 digits and these 4 digits will get prepended with a 5? Will this take care of all incoming patterns?

Hope this helps.

Brandon

David Hailey · ‎04-19-2010

This will actually depend on your SRST configurations. When the phones lose connectivity to the CUCM and go into SRST mode, they will use the SRST reference assigned to them in their DP and attempt to register with that gateway. From there, they will register as ephones with the router. As far as inbound/outbound, this depends on how you configure the router and the capacity you have on your SIP trunk - i.e., can you accomodate inbound calls to everyone or only some folks, do you want to provide outbound only, do you want outbound for everyone but inbound for some VIP's? Those things are mostly up to you.

Hailey

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