[Asterisk-Users] SIP call failover

[Damon Estep]

I send SIP traffic to two different providers.

 

My intention is to try the lower cost provider first, and if there is no
response then send the call to the alternate provider.

 

It seems that if the first provider is having issues the call does not
progress to the second provider, possibly a timeout issue?

 

The goal is to wait about 5 seconds for call progress from SIP provider
A, and if there is noprogress send the call to SIP provider B.

 

Can anyone shed some light on how to timeout the call faster if there is
no progress (without timing out the ring time after progress)?

 

Here are the relevant parts of the configs;

 

Dialplan macro;

 

<snip>

exten => s,6,dial(SIP/${MACRO_EXTEN:1}@sip_a,60)

exten => s,7,dial(SIP/${MACRO_EXTEN:1}@sip_b,60)

<snip>

 

Sip.conf

 

[sip_a]

type=peer

host=x.x.x.x

context=subscriber

dtmfmode=inband

canreinvite=no

 

[sip_b]

type=peer

host=x.x.x.x

context=subscriber

dtmfmode=inband

 

 

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