SIP Trunk Incoming Call Failover

Asterisk Asterisk Support
1

Hi Guys,

I have an Asterisk server passing calls to an Avaya Communication Manager. One SIP trunk is registered to one Avaya CLAN while another trunk is registered to a different CLAN.

I’m trying to failover incoming calls if one trunk is unreachable.

Below is from my extensions.conf setup

exten=>_05xxxxxx,1,Answer
exten=>n,Set(NETWORKSTATUS=${SIPPEER(AvayaTrunk2,status)})
exten=>n,Gotoif($[NETWORKSTATUS=UNREACHABLE]?101)
exten=>n,Dial(SIP/1097@AvayaTrunk2)
exten=>101,Dial(SIP/1097@AvayaTrunk)

When I dial the 05 number I receive the below error - am I missing something?

-- Executing [05xxxxxx@sipprovider:1] Answer("SIP/sipprovider-0000000d", "") in new stack
-- Auto fallthrough, channel 'SIP/sipprovider-0000000d' status is 'UNKNOWN'

Thanks for any help

2

You need a pattern that matches the extension number on every line, or you need to use “same”.

3

thanks - I’ve made it a little further - I have my AvayaTrunk2 in an unreachable state -

AvayaTrunk2 10.0.20.39 a 5060 UNREACHABLE

However when I call in now it doesn’t route to AvayaTrunk which is available as per the gotoif

-- Executing [05xxxxxx@sipprovider:1] Answer("SIP/sipprovider-00000012", "") in new stack
-- Executing [05xxxxxx@sipprovider:2] Set("SIP/sipprovider-00000012", "NETWORKSTATUS=UNREACHABLE") in new stack
-- Executing [05xxxxx@sipprovider:3] GotoIf("SIP/sipprovider-00000012", "0?101") in new stack
-- Executing [05xxxxx@sipprovider:4] Dial("SIP/sipprovider-00000012", "SIP/1097@AvayaTrunk2") in new stack

[Aug 21 16:59:28] WARNING[16211][C-00000015]: app_dial.c:2437 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)

My modified code as per your same recommendation:

exten=>_05xxxxxx,1,Answer
same=>n,Set(NETWORKSTATUS=${SIPPEER(AvayaTrunk2,status)})
same=>n,Gotoif($[NETWORKSTATUS=UNREACHABLE]?101)
same=>n,Dial(SIP/1097@AvayaTrunk2)
same=>101,Dial(SIP/1097@AvayaTrunk)

Any ideas at all? thanks a lot

4

That normally results when a dynamic peer fails to register, but you would normally use a static address in this case.

5

yes both peers are on static addresses - I have purposely put AvayaTrunk2 into an UNREACHABLE state to test the failover to AvayaTrunk. Any ideas as to why it is not failing over when I dial? Thanks again!

6

Because the qualify poll hasn’t been done since you created the fault. In general, you should do the dial and then handle the failure, rather than try to pre-empt the result of the dial.

In this case, I suspect that the qualify was done between the test and the Dial, as I have a feeling that a timeout on the Dial woudl produce a different error, but I could be wrong.

7

Actually, there is an error in the expression. You are comparing the string NETWORKSTATUS against the string UNREACHABLE, which will never be true, rather than comparing a variable against a string.

This doesn’t invalidate the fact that you should try the Dial and analyze the error, if you want to avoid race conditions

8

I got this working thanks for your help - as you said it was best to dial the peer first and then catch the error. I tweaked the code slightly also changing to dialstatus and channel unavailable.

Below is the working code in case anyone else is interested - this receives a call on AvayaTrunk2 and if not available tries AvayaTrunk instead.

exten=>_05xxxxxx,1,Dial(SIP/1097@AvayaTrunk2)
same=>n,Gotoif($[${DIALSTATUS}=CHANUNAVAIL]?continue)
Same=>n(continue),Dial(SIP/1097@AvayaTrunk)

:smile: