[cisco-voip] intercluster trunk over IPSec VPN

Abebe Amare abucho at gmail.com
Thu Feb 9 04:52:50 EST 2012


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/<http://www.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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