Outbound calls fail when trunk registers to secondary sip-server

[jgraylewis]

Verified that the trunk registers to the secondary Sip-Server when the primary is unavailable, however; outbound calls are still directed to the primary and subsequently fail (404 not found).  Inbound calls complete successfully.

Cisco IAD 2431

C2430-is-mz.124-11.xw7.bin

dial-peer voice 100 voip

destination-pattern xxxx

voice-class codec 711

session protocol sipv2

session target sip-server

sip-ua

no remote-party-id

set pstn-cause 47 sip-status 486

retry invite 2

retry response 3

retry bye 3

retry prack 6

timers expires 300000

registrar dns:FQDN expires 3600

sip-server dns:FQDN

[acampbell]

John,

The issue here looks to be DNS related.

Your dial peer

dial-peer voice 100 voip

destination-pattern xxxx

voice-class codec 711

session protocol sipv2

session target sip-server

points at the SIP-SERVER

The SIP-SERVER is DNS contolled

sip-ua

no remote-party-id

set pstn-cause 47 sip-status 486

retry invite 2

retry response 3

retry bye 3

retry prack 6

timers expires 300000

registrar dns:FQDN expires 3600

sip-server dns:FQDN

You could try just using IP addrsses and 2 dial peers

!

dial-peer voice 100 voip

destination-pattern xxxx

voice-class codec 711

session protocol sipv2

session target ipv4:10.10.10.100

voice-class sip options-keepalive up 30 down 5

peference 1

!

!

dial-peer voice 200 voip

destination-pattern xxxx

voice-class codec 711

session protocol sipv2

session target ipv4:10.10.10.200

voice-class sip options-keepalive up 30 down 5

peference 2

!

This will give preference to the 1st dial peer and conrinually check for the remote end being alive

When it fails the 2nd dial peer would kick in.

HTH

Alex

[Luis Sanchez]

Hey Guys,

Is this still the best way to do this?  I only have a single sip server defined in my configuration and was wondering how to setup a backup sip server in case the primary goes down.

sip-ua

no remote-party-id

retry invite 2

retry register 10

timers connect 100

sip-server ipv4:4.XX.XX.13:5070

!

!

dial-peer voice 450 voip

translation-profile outgoing SIPDIVERSION_OUT

destination-pattern 011T

voice-class sip profiles 1

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

codec g711ulaw

fax-relay ecm disable

fax rate 9600

fax nsf 000000

ip qos dscp cs5 media

no vad

[Ayodeji Okanlawon]

Luis,

There are two ways you can achieve this

1. Use DNS SRV records

2. Multiple dial-peers with prefeences

With DNS SRV records you can have a single dial-peer that will send calls to multiple SIP servers. You will use DNS SRV records to define the FQDN of these servers

This needs a bit of work to get it going..So the easies option is to use multiple dial-peers...To learn how to setup both method please read this article

http://beaccie.blogspot.co.uk/2009_07_01_archive.html

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