[Asterisk-Users] Application Hangup not hanging up, possible dialplan cockup?

[JP Hindin]

Greetings all;

I have an odd problem - Hangup isn't hanging up, instead Asterisk carries
the flow going in the extensions.conf, and the next matching extension
gets run. Not good. My extensions.conf (highly simplified) looks like
this:

[pri]
include => dids
include => SIPlookup

[dids]
exten => 13015555555,1,Wait,3
exten => 13015555555,2,Answer
exten => 13015555555,3,Playback(im-sorry)
exten => 13015555555,4,Playback(nbdy-avail-to-take-call)
exten => 13015555555,5,Playback(carried-away-by-monkeys)
exten => 13015555555,6,Playback(tt-monkeys)
exten => 13015555555,7,Hangup

[SIPlookup]
exten => _.,1,AGI,SomethingElse


If I dial the 13015555555 extension, it carries through like it's supposed
to - and when it hits Hangup, Asterisk's CLI reports it's SUPPOSED to be
hanging up - but it doesn't, and jumps straight into SIPlookup's catch-all
and into the AGI. Not good at all.

I run a GrandStream box, but I've had a friend test this with another SIP
phone to the same results. (Keeping in mind this is Asterisk not hanging
up, rather than the phones not hanging up)

The SIP debug looks a little like this:

    -- Executing Wait("SIP/3015845559-bf64", "3") in new stack

Sip read:
ACK sip:13015555555 at serverIP SIP/2.0
Via: SIP/2.0/UDP clientIP;branch=z9hG4bKc4166a0beedc9579
From: "JP Hindin" <sip:3015845559 at serverIP>;tag=a375d9bac322f73f
To: <sip:13015555555 at serverIP>;tag=as12bc5f06
Contact: <sip:3015845559 at clientIP>
Proxy-Authorization: DIGEST username="3015845559", realm="asterisk", algorithm=MD5, uri="sip:130155$Call-ID: a4a6525be5bd113a at 192.168.0.24
CSeq: 33097 ACK
User-Agent: Grandstream HT486 1.0.4.59
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0


12 headers, 0 lines
    -- Executing Playback("SIP/3015845559-bf64", "im-sorry") in new stack
    -- Playing 'im-sorry' (language 'en')
    -- Executing Playback("SIP/3015845559-bf64", "nbdy-avail-to-take-call") in new stack
    -- Playing 'nbdy-avail-to-take-call' (language 'en')
    -- Executing Playback("SIP/3015845559-bf64", "carried-away-by-monkeys") in new stack
    -- Playing 'carried-away-by-monkeys' (language 'en')
    -- Executing Playback("SIP/3015845559-bf64", "tt-monkeys") in new stack
    -- Playing 'tt-monkeys' (language 'en')
    -- Executing Hangup("SIP/3015845559-bf64", "") in new stack
  == Spawn extension (binhost-staff, 1301555555, 11) exited non-zero on 'SIP/3015845559-bf64'
    -- Executing AGI("SIP/3015845559-bf64", "SomethingElse") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/SomethingElse
    -- AGI Script Executing Application: (Playback) Options: (invalid)
    -- Playing 'invalid' (language 'en')


Which, as you can see, it sends a Hangup - but the SIP debug shows no
actual hanging being sent via SIP. We're running Asterisk v0.90.

Any help or suggestions is greatly appreciated.

Many thanks;
JP