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How to create a "hunt group" in SRST configs?

ajrussell
Level 1
Level 1

I'm curious how this would be done.  The scenario is that I have a branch office with approx. 50 people.  In the event of circuit failure, I would like to configure the SRST to ring a specific order of phones through my FXO ports - let's just say ten for discussion's sake.  Is this as simple as the following?

default-destination XXXX

alias 1 XXXX to 1111

alias 2 XXXX to 1112

alias 3 XXXX to 1113... and so on through alias 10

If not, how would I achieve the above requirement?

Thanks for the help...

6 Replies 6

ajrussell
Level 1
Level 1

I suppose it would be useful to see the rest of the SRST config already on the router:

call-manager-fallback
secondary-dialtone 9
max-conferences 8 gain -6
transfer-system full-consult
timeouts interdigit 6
ip source-address 10.1.1.1 port 2000
max-ephones 50
max-dn 192
system message primary Primary Link Down
system message secondary Primary Link Down
transfer-pattern ....
voicemail 93039871111
moh BeStillMyLove.ulaw.wav
multicast moh 239.1.1.1 port 16384 route 10.1.1.1 10.1.50.1
time-zone 6

Would adding the previously mentioned alias commands provide the functionality I'm after?

With just SRST you cannot, you need CME-as-SRST for this.

SRST Fallback Mode Using Cisco Unified CME

This feature enables routers to provide call-handling support for  Cisco Unified IP phones if they lose connection to remote primary,  secondary, or tertiary Cisco Unified Communications Manager  installations or if the WAN connection is down. When Cisco Unified SRST  functionality is provided by Cisco Unified CME, provisioning of phones  is automatic and most Cisco Unified CME features are available to the  phones during periods of fallback, including hunt-groups, call park and  access to Cisco Unity voice messaging services using SCCP protocol. The  benefit is that Cisco Unified Communications Manager users will gain  access to more features during fallback without any additional licensing  costs.

http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucme/admin/configuration/guide/cmesrst.html

HTH

java

If this helps, please rate

www.cisco.com/go/pdihelpdesk

HTH

java

if this helps, please rate

Joel Hatfield
Level 1
Level 1

Are you using a PLAR on the FXO port for inbound calls?  If all you are looking to do is ring the phones you list in the  specific order then try this:

no huntstop

alias 1 XXXX to 1111 preference 1

alias 2  XXXX to 1112 preference 2

alias 3 XXXX to 1113 preference 3... and so on through alias 10

Thanks, Joel - I believe that is exactly what I need.  Appreciate the help...

Hello all. I am having the same issue here with an MGCP gateway with FXO port.
here is my config under fallback
call-manager-fallback
max-conferences 6 gain -6
transfer-system full-consult
timeouts interdigit 6
ip source-address 10.42.173.1 port 2000
max-ephones 12
max-dn 24
system message primary Service Interruption
transfer-pattern T
keepalive 20
voicemail 290000
no huntstop
pickup 603890
alias 1 294013 to 760230
alias 2 294013 to 760231
alias 3 294013 to 760232
alias 4 294013 to 760228
alias 5 294013 to 760229
call-forward pattern T
call-forward busy 290000
call-forward noan 290000 timeout 15
time-zone 21
time-format 24
date-format dd-mm-yy
cor incoming group4 default

 

I used THESE aliases because in CUCM the 760228, 29, 30 etc were in the huntgroup along with the pilot of 294013


 

this still doesn't work though as this hunt group need to be called from the PSTN so someone dials 01XXX XXX228 which presunably is translated in cucm to 294013 the only tranbslation on the gateway i can see is
voice translation-rule 1
rule 1 /^.*\(....$\)/ /\1/
!
voice translation-rule 2
rule 1 /^.*$/ /90\0/
!
!
voice translation-profile PSTN-Inbound-Translate
translate calling 2
translate called 1
!

 

Do i not need a specific dial-peer as well. or have i added the wrong number to the aliases?

In debug i can see it hits dial-peer 999000 is this correct ? Here is the config gotr that dial peer
!
dial-peer voice 999000 pots
service mgcpapp
port 0/1/0
!
dial-peer voice 999001 pots
service mgcpapp
port 0/1/1
!
dial-peer voice 1 pots
translation-profile incoming PSTN-Inbound-Translate
incoming called-number .
direct-inward-dial

 

 

Two things, first of you have to replace the translations you do in CM into the gateway. This would be the number one reason for why many have abandoned MGCP as it adds to the administration overhead to maintain the configuration in multiple devices and places. Second the alias commands needs to point to directory numbers that exist in SRST. From your description this does not seem to be the case here.



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