10-10-2011 04:49 AM - edited 03-16-2019 07:24 AM
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
10-10-2011 06:48 AM
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
06-24-2013 11:17 AM
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
06-24-2013 03:01 PM
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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