Everything 'sent to invalid extension'

[davidiwharper]

Happy new year S

I am setting up a new V6.0-51 PBX with a new SIP channel and a legacy DAHDI inbound. The system is very simple (I think), and it is a vanilla install on top of Debian 9.

Having set up the dahdi trunk and a few bits and pieces, the most unusual thing is happening. Anything where SAIL is supposed to take over and route to some internal resource (call group, IVR) is failing with a "sent to invalid extension" error.

So here is an internal call to our 'reception' call group 500, which works until it is supposed to resolve by going to our IVR 510:

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    -- Executing [500@internal:1] AGI("SIP/104-00000000", "sarkhpe,OutCos,500,,") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/sarkhpe
    -- <SIP/104-00000000>AGI Script sarkhpe completed, returning 0
    -- Executing [500@104closedcos:1] AGI("SIP/104-00000000", "sarkhpe,OutCluster,500,,") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/sarkhpe
    -- <SIP/104-00000000>AGI Script sarkhpe completed, returning 0
    -- Executing [500@qrxvtmny:1] AGI("SIP/104-00000000", "sarkhpe,Alias,SIP/101&SIP/102&SIP/103,500,") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Dial) Options: (SIP/101&SIP/102&SIP/103,20,ciIkt)
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
    -- Called SIP/101
    -- Called SIP/102
    -- Called SIP/103
    -- Connected line update to SIP/104-00000000 prevented.
    -- Connected line update to SIP/104-00000000 prevented.
    -- Connected line update to SIP/104-00000000 prevented.
    -- SIP/101-00000001 is ringing
    -- SIP/102-00000002 is ringing
    -- SIP/103-00000003 is ringing
    -- Nobody picked up in 20000 ms
    -- <SIP/104-00000000>AGI Script sarkhpe completed, returning 0
    -- Channel 'SIP/104-00000000' sent to invalid extension: context,exten,priority=extensions,510,1
    -- Executing [i@extensions:1] PlayTones("SIP/104-00000000", "congestion") in new stack
    -- Auto fallthrough, channel 'SIP/104-00000000' status is 'NOANSWER'


And here is the inbound DAHDI, noting that I have set '500' as the 'Operator' extension so at least that part works.

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-- Starting simple switch on 'DAHDI/1-1'
    -- Executing [s@from-pstn:1] Set("DAHDI/1-1", "chan=1-1") in new stack
    -- Executing [s@from-pstn:2] Set("DAHDI/1-1", "chan=1") in new stack
    -- Executing [s@from-pstn:3] Goto("DAHDI/1-1", "mainmenu,DAHDI1,1") in new stack
    -- Goto (mainmenu,DAHDI1,1)
    -- Channel 'DAHDI/1-1' sent to invalid extension: context,exten,priority=mainmenu,DAHDI1,1
    -- Executing [i@mainmenu:1] Goto("DAHDI/1-1", "extensions,500,1") in new stack
    -- Goto (extensions,500,1)
    -- Executing [500@extensions:1] AGI("DAHDI/1-1", "sarkhpe,Alias,SIP/101&SIP/102&SIP/103,500,") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Dial) Options: (SIP/101&SIP/102&SIP/103,20,ciIkt)
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
    -- Called SIP/101
    -- Called SIP/102
    -- Called SIP/103
    -- Connected line update to DAHDI/1-1 prevented.
    -- Connected line update to DAHDI/1-1 prevented.
    -- Connected line update to DAHDI/1-1 prevented.
    -- SIP/101-00000004 is ringing
    -- SIP/102-00000005 is ringing
    -- SIP/103-00000006 is ringing
    -- Nobody picked up in 20000 ms
    -- <DAHDI/1-1>AGI Script sarkhpe completed, returning 0
    -- Channel 'DAHDI/1-1' sent to invalid extension: context,exten,priority=extensions,510,1
    -- Executing [i@extensions:1] PlayTones("DAHDI/1-1", "congestion") in new stack
    -- Auto fallthrough, channel 'DAHDI/1-1' status is 'NOANSWER'
    -- Hanging up on 'DAHDI/1-1'
    -- Hungup 'DAHDI/1-1'
    -- Starting simple switch on 'DAHDI/1-1'
    -- Executing [s@from-pstn:1] Set("DAHDI/1-1", "chan=1-1") in new stack
    -- Executing [s@from-pstn:2] Set("DAHDI/1-1", "chan=1") in new stack
    -- Executing [s@from-pstn:3] Goto("DAHDI/1-1", "mainmenu,DAHDI1,1") in new stack
    -- Goto (mainmenu,DAHDI1,1)
    -- Channel 'DAHDI/1-1' sent to invalid extension: context,exten,priority=mainmenu,DAHDI1,1
    -- Executing [i@mainmenu:1] Goto("DAHDI/1-1", "extensions,500,1") in new stack
    -- Goto (extensions,500,1)
    -- Executing [500@extensions:1] AGI("DAHDI/1-1", "sarkhpe,Alias,SIP/101&SIP/102&SIP/103,500,") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Dial) Options: (SIP/101&SIP/102&SIP/103,20,ciIkt)
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
    -- Called SIP/101
    -- Called SIP/102
    -- Called SIP/103
    -- Connected line update to DAHDI/1-1 prevented.
    -- Connected line update to DAHDI/1-1 prevented.
    -- Connected line update to DAHDI/1-1 prevented.
    -- SIP/101-00000007 is ringing
    -- SIP/102-00000008 is ringing
    -- SIP/103-00000009 is ringing
    -- <DAHDI/1-1>AGI Script sarkhpe completed, returning 4
  == Spawn extension (extensions, 500, 1) exited non-zero on 'DAHDI/1-1'
    -- Hanging up on 'DAHDI/1-1'
    -- Hungup 'DAHDI/1-1'


I've probably done something accidental to cause this problem, but I have no idea what. Any clues?

Many thanks
David

[sysadmin]

There isn't really any dedicated support for DAHDI in V6.   The generators were not continued from V5 through lack of demand.  Most customers are either using pure SIP or SIP/ISDN gateways these days.   However, I think we can still get you to the picture show.   You can simply create a custom app to catch the DAHDI dialplan.  Something like _DAHDI/X.  should do it, that will catch any channel assignment and you can then  simply send it on to a regular DDI with a GoTo({myddinum},1).

Like this..

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exten=>_DAHDI/X.,1,GoTo(mainmenu,{myddinum},1)


Create a DDI to catch it, doesn't even need to be a real DDI just something to anchor to.

That should get you home.