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Failover between two SIP trunks (CUBE) take long Time Issue

Remon Adel
Level 1
Level 1

Dears ,
we have issue  of long time of failover between two SIP Trunks of two CUBE on CUCM.
Our scenario as below 

CUCM (PUB-SUB) ----SIP TRUNK 1 -----CUBE 1----SIP----Telco (SIP Service UP from telco)
CUCM (PUB-SUB) ----SIP TRUNK 2 -----CUBE 2----SIP----Telco (SIP Service Down )

We set SIP Trunk1 in RG1 , SIP Trunk2 in RG2 .
Normal Case
Standard Route list Contains ( RG1 then RG2 )
when we try outbound calls , calls success with out any delay .

Failover Case
Standard Route list Contains ( RG2 then RG1 ) 
when we try outbound calls , calls success but with high delay about 60 seconds  or more.

we modify these setting under service parameters but without luck 
set Retry Count for SIP Invite to be 2
set SIP Trying Timer  to be 200 

Please Advice
Thanks

1 Accepted Solution

Accepted Solutions

Have you tried these commands on the CUBE.

sip-ua
retry invite 3
timers trying 200

Also, do you have options ping configured on your cubes. This will busy-out
the dialpeers when no service and when cucm sends invite cube will respond
with 503 message immediately without waiting for provider to timeout.

View solution in original post

6 Replies 6

R0g22
Cisco Employee
Cisco Employee
Is the SIP trunk on the secondary CUBE always registered/ACTIVE ? 60 seconds is a huge delay. I doubt it is either CUCM or CUBE.
Can you take a log from the RG2 CUBE ? Also, what happens if you point SIP trunk 2 directly on the RP bypassing SLRG/RG ?

Now , we set RP to point directly to SIP Trunk-2  (Which Cube has no sip service from telco), it takes long time to get fast busy tone about 60 s
Kindly find attached run config and output of below debugs

debug voip ccapi inout

debug ccsip messages

Have you tried these commands on the CUBE.

sip-ua
retry invite 3
timers trying 200

Also, do you have options ping configured on your cubes. This will busy-out
the dialpeers when no service and when cucm sends invite cube will respond
with 503 message immediately without waiting for provider to timeout.

Hello Mohamed ,

Thanks for your support ,
The issue has been fixed after adding these configs 

sip-ua
retry invite 3
timers trying 200

Also Allow keep alive for all dial peers for ISIP

dial-peer voice X voip

voice-class sip options-keepalive

Thanks



Great news. Please mark the question as answered.

I Accepted your solution already
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