JP Hindin
2004-Jul-22 08:35 UTC
[Asterisk-Users] Application Hangup not hanging up, possible dialplan cockup?
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@serverIP SIP/2.0 Via: SIP/2.0/UDP clientIP;branch=z9hG4bKc4166a0beedc9579 From: "JP Hindin" <sip:3015845559@serverIP>;tag=a375d9bac322f73f To: <sip:13015555555@serverIP>;tag=as12bc5f06 Contact: <sip:3015845559@clientIP> Proxy-Authorization: DIGEST username="3015845559", realm="asterisk", algorithm=MD5, uri="sip:130155$Call-ID: a4a6525be5bd113a@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