Sorry I haven’t replied back to the group sooner, I’ve been working on a number 
of other items the last few days.

I wanted to answer this as I saw it flash by before I became sidetracked again.

If you use the options keepalive under the peer, it appears to choose a 
destination to send the SIP options message to. It will send the message to the 
same peer more than once, so I don’t know if it is randomly choosing the same 
one multiple times, or if it only chooses a new primary target every so often. 
All I have had experience with at the moment is that the far end peer was one 
that the SIP trunk wasn’t active on (UCM) and it was replying back with a 
service unavailable message. With enough of these in a row, the whole dial peer 
busies out. It does not attempt to reach other members immediately.

I probably should test this to see if it will attempt to poll another peer if 
the connection (TCP) fails. It probably will not. In this case, I would have to 
rely on the SRV itself and the invite and setup timers to handle the case of 
trying to have redundancy, and nix the options ping as not mixing with using 
SRV records.

I have it in my head that using, say, 4 hosts under the service records will 
allow the router to try another if setup fails (and this works, after the 
timers expire). Setting four peers to priority 0 does not, it fails and hunts 
for the next matching peer or priority. I set it up this way to avoid n^2 
number of peers per matching destination, and to achieve some sort of ‘poor 
man’s load balancing’ even if that isn’t really required.


Adam


From: NateCCIE [mailto:[email protected]]
Sent: Thursday, July 21, 2016 3:32 PM
To: Nick Barnett
Cc: Erick Bergquist; [email protected]; Pawlowski, Adam
Subject: Re: [cisco-voip] Route Dial-Peer Based On Response

I have only used SRV destination a little bit, but my CVP friends use them 
extensively.  I have never seen the option ping with SRV records act 
differently than I would have expected.

Do you have experiences where the whole dialpeer went down and other members of 
the SRV were still accessable?

Sent from my iPhone

On Jul 21, 2016, at 9:11 AM, Nick Barnett 
<[email protected]<mailto:[email protected]>> wrote:
That may work fine based on how the SRV and options pings are configured... but 
if you are counting on an SRV record to point to another CCM sub when the 
primary is down (etc...), options pings will likely take the whole DP down when 
a single host in the SRV goes unreachable.

On Wed, Jul 20, 2016 at 6:12 PM, Erick Bergquist 
<[email protected]<mailto:[email protected]>> wrote:
You could also look at adding keep alive (options ping) to the dial peers and 
call manager sip trunk with options mentioned above.

Do you mind sharing the tcl script?

Erick


On Wednesday, July 20, 2016, Pawlowski, Adam 
<[email protected]<mailto:[email protected]>> wrote:
Nathan,

                Thanks, this looks to be exactly what I’m looking for, this way 
I don’t convey the wrong message. It doesn’t seem like I can have more than one 
option other than hunt or don’t hunt, and it seems to be proper to let the 
telephony provider handle it. Cool. Thanks again.

Adam

From: Nathan Richardson [mailto:[email protected]]
Sent: Wednesday, July 20, 2016 12:34 PM
To: Pawlowski, Adam; '[email protected]'
Subject: RE: Route Dial-Peer Based On Response

One thing that may help is to configure the “voice hunt” settings. For example, 
you could put in “no voice hunt temp-fail” which would make the router stop 
routing if it receives a cause code 41 from the CM so it would skip your TCL 
script in that scenario and should send that code back to your ITSP. It may 
even work to combine “no voice hunt all” with “voice hunt unassigned-number” or 
something like that.

http://www.cisco.com/en/US/docs/ios/12_3t/voice/command/reference/vrht_v2_ps5207_TSD_Products_Command_Reference_Chapter.html#wp1190281

-Nathan Richardson

From: cisco-voip [mailto:[email protected]] On Behalf Of 
Pawlowski, Adam
Sent: Wednesday, July 20, 2016 5:33 AM
To: '[email protected]' 
<[email protected]<mailto:[email protected]>>
Subject: [cisco-voip] Route Dial-Peer Based On Response

[External Email]
Hey all,

                I’ve set up our CUBE routers to try and be a bit more slick, so 
I am making use of e164 pattern maps, dial peer groups, and DNS SRV lookups for 
redundancy/randomization. All that actually seems to be working rather well. I 
have a requirement to make any inactive/unallocated number in my UCM play a 
custome intercept. I did this, at least for now, by setting up a secondary dial 
peer that matches with a higher preference than my UCM peer, and it plays an 
announcement with a TCL script.

                I’d like to set this up so that if the UCM peer is down, or if 
it receives some other code indicating a temporary failure, etc, I either would 
like to bypass this peer so the code goes back to the ITSP, or I can play a 
message saying something about technical difficulties, etc. I’m not sure it’s 
possible to do this? The other way of doing this would be to have the UCM 
itself with a translation or something to roll to an audiotext mailbox, which 
is how we do this today, but it requires either that we maintain translations 
for all numbers, or a generic one that will answer to all extensions queried at 
the system which I don’t want to do either.

                Any thoughts?

Regards,

Adam Pawlowski
SUNY Buffalo NCS


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