Suddenly, call flow routing doesn't work thus no incoming call

General Help
1

My inbound route points to call flow control 0 (there is a second one configured)
This morning , suddenly, no incoming call are possible, with voice playing ss-noservice message.
To test, I have configured a third call flow control (2) identical to the first one and made inbound route to point to
This way, all is working correctly

These are the difference in call trace :

Failing one:

-- Executing [437123123@from-trunk:18] Goto("SIP/Mediatrix1-00000006", "app-daynight,1") in new stack
-- Goto (from-trunk,app-daynight,1)
-- Executing [app-daynight@from-trunk:1] Set("SIP/Mediatrix1-00000006", "__FROM_DID=app-daynight") in new stack
-- Executing [app-daynight@from-trunk:2] NoOp("SIP/Mediatrix1-00000006", "Received an unknown call with DID set to app-daynight") in new stack
-- Executing [app-daynight@from-trunk:3] Goto("SIP/Mediatrix1-00000006", "s,a2") in new stack
-- Goto (from-trunk,s,2)
-- Executing [s@from-trunk:2] Answer("SIP/Mediatrix1-00000006", "") in new stack
   > 0x7f0fd80320e0 -- Probation passed - setting RTP source address to 192.168.0.39:5004
-- Executing [s@from-trunk:3] Log("SIP/Mediatrix1-00000006", "WARNING,Friendly Scanner from 192.168.0.39") in new stack

[2016-06-21 11:14:53] WARNING[3040][C-00000002]: Ext. s:3 @ from-trunk: Friendly Scanner from 192.168.0.39
– Executing [s@from-trunk:4] Wait(“SIP/Mediatrix1-00000006”, “2”) in new stack
– Executing [s@from-trunk:5] Playback(“SIP/Mediatrix1-00000006”, “ss-noservice”) in new stack
– <SIP/Mediatrix1-00000006> Playing ‘ss-noservice.slin’ (language ‘it’)
– Executing [s@from-trunk:6] SayAlpha(“SIP/Mediatrix1-00000006”, “app-daynight”) in new stack
– <SIP/Mediatrix1-00000006> Playing ‘letters/a.slin’ (language ‘it’)
– <SIP/Mediatrix1-00000006> Playing ‘letters/p.slin’ (language ‘it’)
– <SIP/Mediatrix1-00000006> Playing ‘letters/p.slin’ (language ‘it’)
– <SIP/Mediatrix1-00000006> Playing ‘letters/dash.slin’ (language ‘it’)
– <SIP/Mediatrix1-00000006> Playing ‘letters/d.slin’ (language ‘it’)
– <SIP/Mediatrix1-00000006> Playing ‘letters/a.slin’ (language ‘it’)
– <SIP/Mediatrix1-00000006> Playing ‘letters/y.slin’ (language ‘it’)
– <SIP/Mediatrix1-00000006> Playing ‘letters/n.slin’ (language ‘it’)
– <SIP/Mediatrix1-00000006> Playing ‘letters/i.slin’ (language ‘it’)
– <SIP/Mediatrix1-00000006> Playing ‘letters/g.slin’ (language ‘it’)
– <SIP/Mediatrix1-00000006> Playing ‘letters/h.slin’ (language ‘it’)
– <SIP/Mediatrix1-00000006> Playing ‘letters/t.slin’ (language ‘it’)

Then hang

Working one :

– Executing [437123123@from-trunk:18] Goto(“SIP/Mediatrix1-00000022”, “app-daynight,2,1”) in new stack
– Goto (app-daynight,2,1)
– Executing [2@app-daynight:1] GotoIf(“SIP/Mediatrix1-00000022”, “0?app-announcement-4,s,1:app-daynight,1,1”) in new stack
– Goto (app-daynight,1,1)
– Executing [1@app-daynight:1] GotoIf(“SIP/Mediatrix1-00000022”, “0?app-announcement-5,s,1:timeconditions,1,1”) in new stack
– Goto (timeconditions,1,1)

after timeconditions there is an IVR as destination

*** Note the first three rows in each situation :

– Executing [437123123@from-trunk:18] Goto(“SIP/Mediatrix1-00000006”, “app-daynight,1”) in new stack
– Goto (from-trunk,app-daynight,1)
– Executing [app-daynight@from-trunk:1] Set(“SIP/Mediatrix1-00000006”, “__FROM_DID=app-daynight”) in new stack

– Executing [437123123@from-trunk:18] Goto(“SIP/Mediatrix1-00000022”, “app-daynight,2,1”) in new stack
– Goto (app-daynight,2,1)
– Executing [2@app-daynight:1] GotoIf(“SIP/Mediatrix1-00000022”, “0?app-announcement-4,s,1:app-daynight,1,1”) in new stack

Shouldn’t the first “app-daynight,1” be “app-daynight,0,1” ???

Why these differences and why without touch anything ??

1 Like
2

I hate to simply say, “And me” but we have just found the same problem. We have two daynight settings. The 1st one dumps caller sin a blackhole while the 2nd one works fine.
Mike

FreePBX 13.0.137

3

Same issue here.
I had to downgrade the framework to version 13.0.131 to have it working again.

1 Like
4

13.0.137 me too

No problem for me to keep the new call flow in meanwhile…

A “curious” thing often happens, after a reboot, framework can’t connect to asterisk (or asterisk not running?)

I MUST issue fwconsole ma upgrade framework and reboot in order to have a working system.

Is this known ?

5

Yes it was known. It was a bug in framework that was fixed yesterday.

1 Like
6

BTW did you notice this message in your log? Did you run a scan locally? Usually friendly scanner is the user agent of sipvicious.

7

^ that was part of the bug. So it’s irrelevant.

8

Andrew, hope this is the correct place to comment this time as it looks like the same issue!

Also ran into this after applying yesterdays updates;

To resolve this same issue I had to change my Call Flow Control Index from the default in the GUI (0) to a
number other than 0 for it to begin working on the inbound routes again.

-Luke

9

Again. This has already been fixed in framework so it doesn’t require any changes.

10

BTW did you notice this message in your log? Did you run a scan locally? Usually friendly scanner is the user agent of sipvicious.

192.168.0.39 is the LAN Mediatrix ISDN gateway to which FreePBX is connected to, so I’m not worried about, should I ??

11

Ignore it like I said earlier.

12

Just upgraded , all is fine :+1: