10-17-2013 10:30 PM - edited 03-16-2019 07:57 PM
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.
10-18-2013 01:19 AM
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.
10-19-2013 08:16 PM
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.
10-21-2013 12:33 AM
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.
10-18-2013 05:53 AM
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
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10-21-2013 03:05 AM
Also test your pots dialpeer by shutting down your 1007 dialpeer and the making an outbound call
Sent from Cisco Technical Support Android App
10-22-2013 02:59 AM
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.
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