cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
6272
Views
10
Helpful
5
Replies

Hunt Group failover in SRST

ty.masse
Level 1
Level 1

I have a remote location with an h.323 gateway that sends call to a hunt group in call manager under normal conditions.  I have two hunt groups.  One rings to the first 4 FXOs and the second rings to the next 4 FXOs from PSTN.

How do I create those hunt groups on the gateway itself when the system is in SRST mode so calls from the outside can reach the same phones that they would reach under normal conditions. 

Thanks.                

1 Accepted Solution

Accepted Solutions

You build seperate Alias ranges, for 5552386 you would have:

alias 1 5552386 to DN1 cfw 5552386 timeout 12

alias 2 5552386 to DN2 cfw 5552386 timeout 12

alias 3 5552386 to DN3 cfw 5552386 timeout 12

alias 4 5552386 to DN4 cfw 5552386 timeout 12

for 4442356 you would continue with alias 5:

alias 5 4442356 to DN1 cfw 4442356 timeout 12

alias 6 4442356 to DN2 cfw 4442356 timeout 12

alias 7 4442356 to DN3 cfw 4442356 timeout 12

alias 8 4442356 to DN4 cfw 4442356 timeout 12

HTH,

Chris

View solution in original post

5 Replies 5

Chris Deren
Hall of Fame
Hall of Fame

Build alias under SRST, here is an exmaple:

calls arrive to 6522 and are hunted to 4 different extensions

alias 1 6522 to 6526 cfw 6522 timeout 12

alias 2 6522 to 6525 cfw 6522 timeout 12

alias 3 6522 to 6524 cfw 6522 timeout 12

alias 4 6522 to 6523 cfw 6522 timeout 12

HTH,

Chris

Thanks for your quick reply.  Just to clarify, when a call comes in PSTN has a roll over hunt group set up which will roll to the next line if the first one is busy.  We have two sets of 4 of those.  How will your example work with calls coming in from pstn thru the voice ports?  Below is my voice port config. 

voice-port 0/0/0

connection plar 4442356

description

!

voice-port 0/0/1

connection plar 4442356

description

!

voice-port 0/0/2

connection plar 4442356

!        

voice-port 0/0/3

connection plar 4442356

description

!        

voice-port 0/1/0

connection plar 5552386

description 

!

voice-port 0/1/1

  connection plar 5552386

description 

!        

voice-port 0/1/2

  connection plar 5552386

description 

!

voice-port 0/1/3

  connection plar 5552386

  description

voice-port 0/0/0

connection plar 4442356

description

!

voice-port 0/0/1

connection plar 4442356

description

!

voice-port 0/0/2

connection plar 4442356

!        

voice-port 0/0/3

connection plar 4442356

description

!        

voice-port 0/1/0

connection plar 5552386

description 

!

voice-port 0/1/1

  connection plar 5552386

description 

!        

voice-port 0/1/2

  connection plar 5552386

description 

!

voice-port 0/1/3

  connection plar 5552386

  description 

You build seperate Alias ranges, for 5552386 you would have:

alias 1 5552386 to DN1 cfw 5552386 timeout 12

alias 2 5552386 to DN2 cfw 5552386 timeout 12

alias 3 5552386 to DN3 cfw 5552386 timeout 12

alias 4 5552386 to DN4 cfw 5552386 timeout 12

for 4442356 you would continue with alias 5:

alias 5 4442356 to DN1 cfw 4442356 timeout 12

alias 6 4442356 to DN2 cfw 4442356 timeout 12

alias 7 4442356 to DN3 cfw 4442356 timeout 12

alias 8 4442356 to DN4 cfw 4442356 timeout 12

HTH,

Chris

Excellent Chris.  That's awesome.  I'll configure it and give it a go.  That's one more thing in my VOIP tool bag.

Thanks.

I have similar issue  in one branch wr we have CUE . centralized with srst h323 in remote branch , where calls from pstn arrive through fxo and hit an ivr script to forward to hunt group in cucm after prompt play .

Normal condition its working fine but in srst after ivr prompt play call ends .Please advice a soultion to fix this one.