09-24-2012 07:34 AM - edited 03-16-2019 01:21 PM
i'm trying to get 3 7961's to failover to the router that is controlling them and here is our setup in a nutshell
the network is all over the place, we do have layer 2 dot 1q trunks between every switch from the core, and we have a 3750 metro doing QOS and our metro wan to our remote sites. we're trying to get SRST working on three phones as we have not been sucuessfull in the past. i have read a little on the subject, and here is what we have setup as we speak.
call managers: we setup a Device pool with the same configuration as the current DP but i setup a SRST reference in this pool.
router down in the dispatch center:
here is our CM fallback config
call-manager-fallback
secondary-dialtone 9
max-conferences 4 gain -6
transfer-system full-consult
timeouts interdigit 4
ip source-address 172.21.0.17 port 2000 (this is the router ip add)
max-ephones 24
max-dn 24
system message primary SRST Fallback Active
system message secondary SRST Fallback
keepalive 15
we have our dial peers setup for this router, i do not have any Ephones setup, do i need to set them up or will the router get the current config from the phone?
09-24-2012 11:18 AM
So, what is not working? The phones do not register or register but cannot call anywhere?
I take it your SRST reference in CUCM references 172.21.0.17, correct?
Chris
09-24-2012 11:37 AM
more so if my config is ok, currently everything is in production and it is extremely difficult to test with public saftey in factor.
and yes, chris, i do have the ip address in the CUCM
09-24-2012 12:00 PM
Yes, your config is OK, so phones should register OK, whether calls can be made depends on dial-peer configuration.
HTH, please rate all useful posts!
Chris
09-24-2012 12:13 PM
anything in particular i need to do to the dial peers? our lines are plared and source to the primary call manager on the router.
09-24-2012 01:17 PM
Well, you did not show what you already have in place.
Is the GW under normal condition using MGCP, SIP or H323?
If SIP or H323 your existing dial-peers will be used under SRST.
If MGCP you will need to build a dial-peer for every pattern you need there, i.e. emergency, local, LD, international pointing to proper voice ports.
HTH, please rate all useful posts!
Chris
09-25-2012 05:56 AM
whoops i forgot to add those.
voice port config
voice-port 0/0/0
no battery-reversal
no vad
connection plar opx immediate 2100
description 2100
caller-id enable
!
voice-port 0/0/1
no vad
connection plar opx immediate 2101
description 2101
caller-id enable
!
voice-port 0/0/2
no vad
connection plar opx immediate 2637
description 911 line 1
caller-id enable
!
voice-port 0/0/3
no vad
!
voice-port 0/1/0
no vad
connection plar opx immediate 2102
description 2102
caller-id enable
!
voice-port 0/1/1
no vad
caller-id enable
!
voice-port 0/1/2
no vad
connection plar opx immediate 2913
description 911 line 2
caller-id enable
!
voice-port 0/1/3
no vad
connection plar opx immediate 2899
description 911 line 2
caller-id enable
mgcp fax t38 ecm
mgcp behavior g729-variants static-pt
!
!
!
dial-peer voice 90911 pots
destination-pattern 9911
port 0/0/0
forward-digits 3
!
dial-peer voice 91911 pots
preference 1
destination-pattern 9911
port 0/0/1
forward-digits 3
!
dial-peer voice 92911 pots
preference 2
destination-pattern 9911
port 0/0/2
forward-digits 3
!
dial-peer voice 93911 pots
preference 3
destination-pattern 9911
port 0/0/3
forward-digits 3
!
dial-peer voice 9500 pots
preference 5
destination-pattern 9.T
port 0/0/0
!
dial-peer voice 9501 pots
preference 4
destination-pattern 9.T
port 0/0/1
!
dial-peer voice 9502 pots
preference 3
destination-pattern 9.T
port 0/0/2
!
dial-peer voice 9503 pots
preference 2
destination-pattern 9.T
port 0/0/3
!
dial-peer voice 401 voip
preference 1
destination-pattern [0-8]...
session target ipv4:172.21.2.10
dtmf-relay h245-signal
codec g711ulaw
no vad
!
dial-peer voice 100 pots
description **Incoming dial peer
incoming called-number 2...
no digit-strip
!
dial-peer voice 402 voip
destination-pattern [0-8]...
session target ipv4:172.21.0.10
dtmf-relay h245-signal
codec g711ulaw
no vad
on the normal gateway we're using H323 thankfully.
10-12-2012 07:05 AM
ok chris, here is what i discovered when we took down the core and had our pub and sub down.
SRST worked just as intended, however, dispatchers were not able to dial out, we have a dial code of 9 to get out. do we need to setup a dial plan on the router?
second of all, once they're in SRST they stay in SRST, in order to get them to fall back to the primary call manager the phone needs reset. is that normal, or should it fail back to the pub or sub when they see connectivity?
10-12-2012 07:12 AM
You need proper dial peer to match what is being dialed.
Phones will re-register automatically , by default the timer is 2 minutes and is defined under device pool --> Connection Monitor Duration. Did you wait 2 minutes?
HTH, please rate all helpful posts!
Chris
10-12-2012 07:15 AM
i dont recall if i waited 2 mintues or not, by the time someone else looked at it they said "huh, didnt go back" and restarted the phone. i will check on that when i can on monday.
regarding the dial peers, since they are pots lines, do i have to take a look at the pots peers?
10-12-2012 07:20 AM
Yes, when you are under SRST use "debug voice dialpeer" to troubleshoot if they are being matched properly.
Chris
10-13-2012 01:07 AM
An easier way, and much less intrusive for other services, to force the site into SRST would be to add specific host routes that point to null for the CUCM IP addresses on the egress gateway. With this you would only affect the phones ability to communicate with the CUCM, not shutting down the rest of communication to/from the site.
A benefit of this is that you will still be able to communicate with the VGW used for SRST, so you can test and verify SRST without being on site. You would of course still need some one local at the site to do some test calls from the physical phone.
Also consider use of trunk group(s) instead of duplicate dial peers. That would be a lot cleaner and make your config more readable.
Please rate all useful posts.
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