11-26-2010 03:43 PM - edited 03-16-2019 02:08 AM
Hi,
I am using a 2821 one as a gateway to route PSTN PRI calls to/from my SIP enabled PBX. In the even that my PBX is not available, I need to route my PSTN calls to a different PBX. I am using preferenced dial-peers to do so. The problem I am experiencing is that the roll over to the next dial-peer is not fast enough. The PSTN will drop my call before the 3rd (Or even the 2nd) dial-peer can setup the SIP connection.
I need to either:
a. - have my dial-peers monitor their session target and place them self in a "down" state when the target is not reachable
b. - find a way to have the dial-peers time out to the next preference much faster than they currently do.
c. - Have the gateway preliminarily answer the call and provide some sort of ring back to the Telco while it runs through the dial-peers to find an operative choice. (this one is probably not gonna happen)
I have reasearched and found some solutions which almost work, such as call fall back. But from what I've read on call fall back, it seems to only work if you're monitoring another cisco device which is configured to respond to your probes. Which I'm not.
My gateways perform very basic call routing, with everything is is handled by the PBX. I don't have SRST, but I will get it if someone can tell me it will solve my problems. Here is an example of 3 dial-peers I have setup via preference.
dial-peer voice 1 voip
description All 5555555 Calls to PBX A
preference 1
destination-pattern 5555555
voice-class codec 1
session protocol sipv2
session target ipv4:10.10.10.10
session transport tcp
dtmf-relay rtp-nte
no vad
!
dial-peer voice 2 voip
description All 5555555 Calls to PBX B
preference 2
destination-pattern 5555555
voice-class codec 1
session protocol sipv2
session target ipv4:10.10.10.11
session transport tcp
dtmf-relay rtp-nte
no vad
!
dial-peer voice 3 voip
description All 5555555 Calls to PBX C
preference 3
destination-pattern 5555555
voice-class codec 1
session protocol sipv2
session target ipv4:10.10.10.12
session transport tcp
dtmf-relay rtp-nte
no vad
Anyone have any ideas of what I can do here?
Thanks,
Dave
11-26-2010 04:47 PM
Under sip-ua, try timer. I think should be "trying".
11-28-2010 11:35 AM
The Out-of-Dialog OPTIONS Ping Failure feature may resolve your issue - more details at the link below.
12-14-2010 09:31 AM
Hi,
I'm not sure if it is normal procedure to reply to your own post with a solution, but here goes...
First I setup some IP SLA monitoring on my primary and secondary phone servers:
!
ip sla 201
icmp-echo 10.x.x.1 source-interface Loopback1
!
ip sla 202
icmp-echo 10.x.x.2 source-interface Loopback1
!
ip sla group schedule 200 201-202 schedule-period 2 frequency 10 start-time now life forever
! The group schedule is not necessary. Having my monitoring synced up just satisfies my anal retentive tendencies
Then I setup some tracking on the SLA objects:
!
track 201 rtr 201 reachability
!
track 202 rtr 202 reachability
!
track 200 list boolean or
object 201
object 202
delay down 30 up 30
Then I used EEM to change the dialpeer prefrence based on my track list:
event manager applet PhoneServersBOTHdown
event track 200 state down
action 0 cli command "enable"
action 1 cli command "conf t"
action 2 cli command "dial-peer voice 555 voip"
action 3 cli command "pref 10"
action 4 cli command "end"
action 5 syslog priority errors msg "10.x.x.1 and 10.x.x.2 unpingable; change dialpeer 555 pref 10"
!
event manager applet PhoneServerUP
event track 200 state up
action 0 cli command "enable"
action 1 cli command "conf t"
action 2 cli command "dial-peer voice 555 voip"
action 3 cli command "pref 1"
action 4 cli command "end"
action 5 syslog priority errors msg "10.x.x.1 and 10.x.x.1 pingable(recovered); change dialpeer 555 pref 1"
This is by no means the best solution, but as far as I know, it is the only solution if you are not running CUBE or SRST
Cheers,
Dave
01-28-2013 09:22 PM
Hi all.
did someone find some better solution since 2010 ?
11-12-2013 03:33 AM
I have just implemented what James stated earlier and it works like a charm!
This question MUST be marked as ANSWERED.
11-12-2013 05:56 AM
You can adjust the sip timers as follows and that should do the magic for you. This is especially useful if the other party doesnt support options PING
sip-ua
retry invite 2
timers trying 150
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