sip dial-peer failover not working

[kurtdeneef]

HI,

I have an issue with 2 sip voip dial-peers towards a Genesys system. Our gateway (2800 with 12.4(11)Y) is containing the sip dial-peers while the Genesys is H323.

When the first Genesys server goes down completely, so ip address is not reachable anymore then the second dial-peer does not take over. We hear a busy tone. After investigation on the second Genesys server, they don't see any signalling coming in at all. So there must be something on the Cisco router.

This is my config:

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

no supplementary-service h225-notify cid-update

redirect ip2ip

h323

sip

voice class h323 1

  call start fast

  h245 caps mode unrestricted

Dial-peer configuration:

dial-peer voice 1700300 voip

description IP-PHONE:Genesys POC GVP17003XX

destination-pattern 17003..

progress_ind setup enable 3

progress_ind progress enable 8

session protocol sipv2

session target ipv4:153.89.39.103

dtmf-relay rtp-nte

codec g711alaw

no vad

!

dial-peer voice 1700301 voip

description IP-PHONE:Genesys POC GVP17003XX

preference 1

destination-pattern 17003..

progress_ind setup enable 3

progress_ind progress enable 8

session protocol sipv2

session target ipv4:153.89.39.107

dtmf-relay rtp-nte

codec g711alaw

no vad

!

I first thought that the preference would be the issue (first dial-peer is using default preference 0) but i don't think that is causing the failover not to work. Anyway, i will reconfigure them with preference 1 and 2.

Maybe there is a certain (sip) failover timer  i need to modify? or any other parameter missing?Unfortunately, i don't have any traces (ccsip or whatever) as they reported the issue to me after the DRP test they did.

I know there is H323 tcp timout -timer you can configure but i guess that i not applicable in my case.

Ony thoughts on that?

Thanks for any feedback, guys.

Kind regards,

Kurt

[Ayodeji Okanlawon]

Try this..

sip-ua

retry invite 2
timers trying 150

Please rate useful posts

[Mohammed al Baqari]

Hi,

What AO mentioned is going to work. I just wanted to add more clarity on why its not working.

Default SIP settings won't allow dial-peer failover to take place. The default number of SIP INVITE retries is 6 while the initial TRYING timer is 500 msec. In you case when the call arrive:

1. Initial SIP invite request
2. 1st invite retry (delay to retry: ~500ms)
3. 2nd invite retry (delay to retry: ~1sec)
4. 3rd invite retry (delay to retry: ~2sec)
5. 4th invite retry (delay to retry: ~4sec)
6. 5th invite retry (delay to retry: ~8sec)
7. 6th invite retry (delay to retry: ~16sec)
8. Failover time to second UAS      (delay to failover: ~32)

Note: Time delay increases using the formula (2xOld Time)

Therefore failover will take place after 63.5 sec. Definitly, by this time the call will be abonded. Therefore you need to reduce the timer.

Based on the above settings, the failover will take place in 1.2 sec.

Hope you are clear now . Please rate if you find the post useful

[Ayodeji Okanlawon]

Mohammed +5 for this excellent explanation! Really Nice!

Please rate useful posts

[kurtdeneef]

Allright, Mohammed.

Thanks a lot for the explanation. I gave you a 5 star rating as this is explained very clear and saves me some research work

nevertheless, one more question, you say "

Time delay increases using the formula (2xOld Time)". What do you mean exactly? i don't see why you would multiply by 2?

Thanks a lot

Kurt

[kurtdeneef]

ok, i see what you mean. delays are doubled each time.

read the post a bit too fast.

Thanks again,

Kurt

[kurtdeneef]

Thank you also for your feedback.

I mark it as a "correct answer" after my test on thursday evening ;-)

Cheers,

Kurt

[Mohammed al Baqari]

Thanks to both of you for your nice words. Glad to know that his helps

[Israel Vargas]

Hi help me please. i have a problem whit the FAC (Force Autoritation Code). I set the dial-peer.  However, when digit number, just take de Dial-Peer  101 Voip, asked the ID number and then the PIN number but, after dialing the key #, the call is cut.

dial-peer voice 101 voip

corlist outgoing call-CELULAR-LOCAL

preference 1

service clid_authen_collect

destination-pattern 9044..........

voice-class codec 1

session target ipv4:10.1.1.1

incoming called-number 9044..........

dtmf-relay h245-alphanumeric

no vad

ial-peer voice 51 pots
corlist outgoing call-CELULAR-LOCAL
preference 2
destination-pattern 9044..........
port 0/1/1
forward-digits 13
no sip-register

I put a "Show voice call status" and showme this:

UC520#sh voice call status

CallID     CID  ccVdb      Port             DSP/Ch  Called #   Codec    MLPP Dial-peers

0x36A      1F0B 0x872DF234 50/0/11.0                5515125936 g711ulaw       20006/101

1 active call found

Namely take the dial-peer 101  but not take the dial-peer 51, and this is the port where the call goes out.

Thanks for help me.