Foip question

[safety2008]

Good morning.

This may be a stupid question but i must ask. We have a quadrant fax solution in place today. Made up of a primary and slave server. each one is connected via a 3845 GW by t1. now if one fax server goes down the slave would pick up the traffic, same if one of the 3845's went down the other would recieve the traffic.

My question is they are proposing a VM soulution of FOIP, Dialogic sr140 technology. I am having a hard time grasping how fail over would work. Since now the dial peers point to a port within their own GW..I assume now the peers will be directed at an IP, And should the VM fax server fail, how would you direct traffic to the secondary IP within the 3845??

any adice thoughts would be appreciated.

Thank you

[Jonathan Schulenberg]

Create two VoIP dial-peers, preferrably over UDP (TCP adds delay during a failure since the TCP handshake needs to timeout as well as the SIP INVITE) but either will work. When the router doesn't receive a response to the SIP INVITE it'll proceed to the next dial-peer match.

sip-ua
  retry invite 2
  ! Default: 6
  ! IOS will only try another dial-peer if this is set to <= 4

This delay will happen while it waits for the first dial-peer will happen on every call. If the Dialogic supports SIP OPTIONS PING you could add this command to the dial-peer:

voice-class sip options-keepalive

This will send keepalives and when the Dialogic stops responding IOS will busyout the dial-peer until it gets an answer. This way calls will immediately skip to the second dial-peer.

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

[safety2008]

Jonathan,

many thanks as always..dreaded dial peers. my router experience lacks so...so i will as i do now have a dial peer pots, and then i would have 2 entries,dil peer voip, each different ip's, in each router to achieve failover??

Thank you again, and this must be a common configuration for failover ??

[Jonathan Schulenberg]

Dial-peers aren't so bad. They're much easier to teach than building out CUCM.

Here's an example (copy/pasted with small tweaks to make it generic):

trunk group PSTN-Default
description PSTN-facing voice ports
!
voice service voip
rtcp keepalive
fax protocol t38 version 3 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
! Version 3 was added in 15.1(1)T but I suggest upgrading to 15.1(4)M5 or 15.2(4)M2.
modem relay nse codec g711ulaw gw-controlled
sip
  bind control source-interface Loopback0 !Or whatever interface you prefer.
  bind media source-interface Loopback0
!
voice class uri 10 sip
host ipv4:X.X.X.X  !IP of your first Dialogic
!
voice class uri 11 sip
host ipv4:y.y.y.y !IP of your second Dialogic
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 ilbc
codec preference 4 g729r8
video codec h264
!
voice-port 0/0/0:23
trunk-group PSTN-Default
voice-port 0/0/1:23
trunk-group PSTN-Default
!
dial-peer voice 1 pots
trunkgroup PSTN-Default
description INCOMING POTS
incoming called-number .
direct-inward-dial
!
dial-peer voice 10 voip
huntstop
destination-pattern [0-8]...
session protocol sipv2
session target ipv4:x.x.x.x
session transport udp
incoming uri via 10
voice-class codec 1 
voice-class sip options-keepalive
dtmf-relay rtp-nte sip-kpml
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 11 voip
huntstop
destination-pattern [0-8]...
session protocol sipv2
session target ipv4:y.y.y.y
session transport udp
incoming uri via 11
voice-class codec 1 
voice-class sip options-keepalive
dtmf-relay rtp-nte sip-kpml
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 30 pots
trunkgroup PSTN-Default
destination-pattern 9[2-9]..[2-9]......
forward-digits 10
!
dial-peer voice 31 pots
trunkgroup PSTN-Default
destination-pattern 91[2-9]..[2-9]......
forward-digits 11
!
dial-peer voice 32 pots
trunkgroup PSTN-Default
destination-pattern 9011T
prefix 011
!
dial-peer voice 33 pots
trunkgroup PSTN-Default
destination-pattern 9[1-9]11
forward-digits 3
!
dial-peer voice 34 pots
trunkgroup PSTN-Default
destination-pattern 90
forward-digits 1
!
dial-peer voice 35 pots
trunkgroup PSTN-Default
destination-pattern 900
forward-digits 2
!
gateway
media-inactivity-criteria rtplib
timer receive-rtp 1200
!
sip-ua
retry invite 2
timers buffer-invite 200 !Only needed if the caller ID name info comes in a separate FACILITY IE.

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

[safety2008]

ok you'll regret answering this post but.

i have attached one of the 3845 configs..since we have a bunch of did's involved..will this be a rebuild, really, of the 3845 dial peers or is there a way to just change the peers to reflect an ip..if we were to take 8890 for example..it has a peer/pots and peer of dial range voip..

would it be delete EVERY entry or modify through. it seems like a lot of configuration and i want to plan accordingly.

thanks again Jonathan