[cisco-voip] intercluster trunk over IPSec VPN

[Ryan Ratliff]

No, I don't think it's there in 6.1.3 and if it was it would show up in the SIP profile.

Incidentally 6.1(3) base is now over 3 years old.  You really should be on a version that was released in this decade.

-Ryan

On Feb 9, 2012, at 4:52 AM, Abebe Amare wrote:

Hi Ryan,

The CUCM version is 6.1.3.1000-16. Is the SIP options ping parameter available in this version? Where would you enable it if it is available?

thanks in Advance,

Abebe

On Wed, Feb 8, 2012 at 8:07 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
What about a SIP trunk with options ping enabled?

-Ryan

On Feb 8, 2012, at 7:05 AM, Abebe Amare wrote:

Hi Dennis,

Configuring a persistent L2L tunnel proved to be very elusive. I settled for running a periodic ping scheduled to keep the tunnel running.

Thanks for your help

Abebe

On Tue, Feb 7, 2012 at 6:16 PM, Dennis Heim <Dennis.Heim at cdw.com> wrote:
I think you answered your own question. IPSEC tunnel’s take time to bring up. Maybe you could tweak some of the VPN negotiating parameters, or create a separate L2 tunnel profile/group just for your voice that is permanent and does not have an inactivity timer.

 

 

Dennis Heim
Senior Engineer (Unified Communications)
CDW  Advanced Technology Services
10610 9th Place
Bellevue, WA 98004

425.310.5299 Single Number Reach (WA)

317.569.4255 Single Number Reach (IN)
317.569.4201 Fax 
dennis.heim at cdw.com
cdw.com/content/solutions/unified-communications/

 

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Abebe Amare
Sent: Tuesday, February 07, 2012 4:10 AM
To: cisco voip
Subject: [cisco-voip] intercluster trunk over IPSec VPN

 

Dears,

I have configured an Inter-Cluster trunk from CUCM to another site with CUCME. There is an IPSec L2L VPN terminating at ASA 5500 firewall on both ends

CUCM --->ASA 5540--->Internet <---ASA 5510<---CUCME

On the ASA,the IPSec tunnel is terminated after 30 minute of inactivity (default) which is causing a problem. When a phone in one site tries to call another phone in the other site there is a noticeable gap before actual conversation is heard over the phone. Once conversation starts, there is no delay or break in audio. Has anyone faced this issue?

best regards,

Abebe


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