CME SIP dial-peer to POTS dial-peer preference failover?

[NicP270591]

Hi guys,

I have a CME router with both SIP dial peers (preference 1) and FXO PSTN dial-peers for failover (preference 2). When SIP is unreachable, you go to dial a call, it hangs on "Ring out" for a minute or so and then returns busy tone/unknown number error. There is no failover to the preference 2 dial peer on the FXO port (same pattern just preference 2 and mapped to a port rather than SIP).

1. Why is there not failover?

2. Can I reduce the hunt time to a few seconds before it cancels SIP and switches to the PSTN peer?

3. Does this even work by default on CME, or will I need SRST on the router?

Nic.

[Andrey Chertolyas]

Hi Nic,

It should work out of the box, you don't have to enable SRST. Please attach the configuration of the router, debug ccsip messages and debug voice ccapi inout for a call.

You can adjust timers in sip-ua section using timers command.

Kind regards,
Andrew C.

[NicP270591]

Hi Andrew,

Config as follows:

dial-peer voice 1007 voip

corlist outgoing callMobile

description IP Mobile Calls

translation-profile outgoing 1

preference 1

destination-pattern 004........

session protocol sipv2

session target sip-server

voice-class codec 1 

dtmf-relay rtp-nte

no vad  

!        

dial-peer voice 2017 pots

corlist outgoing callMobile

description POTS IP Mobile Calls BKCLID

preference 2

destination-pattern 004........

port 0/0/0

forward-digits 14

prefix 1831

no register e164

!        

sip-ua   

credentials username USERNAME password 7 PASSWORD realm REALM

authentication username USERNAME password 7 PASSWORD realm REALM

no remote-party-id

retry invite 2

timers trying 150

registrar dns:sipprovider.net expires 3600

sip-server dns:sipprovider.net

!        

I attempted debug as requested, nothing showed up in my telnet session during a call attempt. Odd.

Nic.

[Andrey Chertolyas]

Hi Nic,

The config seems fine for me, the only thing you can change is to specify the destination-pattern more strictly -

^004........$

It's difficult to say what's wrong without the debug, maybe you forgot to say 'term mon'?

Kind regards,
Andrew C.

[Jonathan Schulenberg]

I'm not saying this is the only problem, but one problem would be the default value of sip-ua > retry invite command. By default the value it has essentially causes a huntstop if all six retries fail. Try changing the command to four or less (I use two) and see if that makes a difference. Beyond that we need the details Andrew asked for.

http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_r1.html#wp1550337

sip-ua

retry invite 2

Please remember to rate helpful responses and identify helpful or correct answers.

[Dennis Mink]

Also test your pots dialpeer by shutting down your 1007 dialpeer and the making an outbound call


Sent from Cisco Technical Support Android App

[NicP270591]

I shut down my 1007 for testing and the call failed. But normally it works fine. Very strange, I will do some testing next time I get the chance and report back. May take a couple days however before I can get on-site.

Nic.