[asterisk-users] Followme Killing Asterisk

A E G all.eforums at gmail.com
Tue Jan 15 11:58:41 CST 2013 

On Tue, Jan 15, 2013 at 11:05 AM, Steve Murphy <murf at parsetree.com> wrote:

> On Mon, Jan 14, 2013 at 9:36 PM, A E G <all.eforums at gmail.com> wrote:
>
>> Hi Guys,
>>
>> this has been a weekend destroyer for me. I've struggled this all day and
>> most of today.
>>
>
> From your discussion below, it sounds like the real problem is the
> Asterisk crashing.
> So, as a first step to solving **that** problem, make sure asterisk is
> compiled with debug
> flags, dumps another core file, and then you do the "gdb asterisk
> <corefilename>", and
> get a stack trace. That should give us some idea of what happened.
>
>
Thanks for the note Steve. It doesn't sound like there's tremendously wrong
that I'm doing as far s the configuration is concerned then? and it won't
be too surprising since the configuration of Followme is quite simple
assuming the complexities are all handled by the Followme app.

I tried a whole lot of options that "made sense" as Dial options that the
"Local" channel dial from Followme is being hooked into but it appears
that, the cause of the crash is most likely that Followme:


   1. Is looking for something to do; bill, log or something after it
   returns from Dial/call termination but not finding it. I tried using
   "Answer(nocdr)" at the time the call on the DID is being answered but that
   didn't help. I have also tried the 'g', 'c', 'C', 'I' and 'i' etc options
   with the Dial but they don't help either. I had real hopes in the 'g'
   option to tell it to proceed with the dial plan where I was simply making
   it return a couple of call status related variables and then just Hangup,
   but regardless of the 'calling' or the called party hanging up, these
   number get printed, which means that despite the 'g' option, the call does
   NOT proceed with the normal/rest of the dialplan
   2.  Maybe Followme is not built for this purpose where the caller is
   unknown (which it would be in most cases) but at least the "called party"
   is usually known AND is a "subscriber"/registered user of the system who is
   then using the Followme feature to find them when they don't answer their
   PBX registered phone. What I'm doing calling from outside, having the
   system answer the call, allow the caller to put in a number and then
   calling those numbers associated with that extension if it's a Followme
   extension but the extension itself isn't a registered user in sip.conf or
   users.conf, and maybe followme app has some procedures it needs to run
   through as a matter of housekeeping (i.e. accounting, billing, logging etc)
   that it's not finding info for

Will do a gdb and see what I can find...I'm not a developer so I may not be
able to pick up a lot from the stack-trace but will pastebin it and see if
one of the community/developer members can figure out why it's taking a dump

Cheers
\a

>
>
>>
>> I have a fairly simple Followme sequence in place to see how it works
>> before I get into the complex scenarios.
>>
>> extensions.conf
>> ---
>> [Incoming]
>> exten => <MyDID>,1, Answer()
>>         same => n, Set(CHANNEL(language)=en_AU)
>>         same => n, Followme(TestFollow)
>>         same => n, NoOp("++Back after Followme: DIALSTATUS =
>> ${DIALSTATUS}, Hangupcause = ${HANGUPCAUSE}")
>>         same => n, Hangup()
>>
>> [Followme-Dialout]
>> exten => _1NXXNXXXXXX,1,Set(CHANNEL(language)=en_AU)
>>         same => n, Dial(SIP/GW-1/${EXTEN})
>>
>> followme.conf
>> ----------------
>> [TestFollow]
>> context => Followme-Dialout
>> number => <my landline>,30
>> number => <my cell phone>,20
>>
>> The call goes out, and rings my first phone. If I answer it, the Asterisk
>> core dumps, the calls stay up!
>>
>> <snip>
>>
>> [Jan 15 04:19:48]     -- Called SIP/GW-1/12035551111
>>
>> [Jan 15 04:19:51]     -- SIP/GW-1-00000007 is making progress passing it
>> to Local/12035551111 at Followme-Dialout-00000004;2
>>
>> [Jan 15 04:19:51]     -- Local/12035551111 at Followme-Dialout-00000004;1
>> is making progress
>>
>> [Jan 15 04:20:05]     -- SIP/GW-1-00000007 answered Local/12035551111
>> @Followme-Dialout-00000004;2
>>
>> [Jan 15 04:20:05]     -- Local/12035551111 at Followme-Dialout-00000004;1
>> answered SIP/DIDProvider-1-00000006
>>
>> [Jan 15 04:20:05]     -- Starting playback of followme/call-from
>>
>> [Jan 15 04:20:05]     -- <Local/12035551111 at Followme-Dialout-00000004;1>
>> Playing 'followme/no-recording.ulaw' (language 'en_AU')
>>
>> [Jan 15 04:20:05]     -- Local/12035551111 at Followme-Dialout-00000004;1
>> requested a source update
>>
>> ast00*CLI>
>>
>> Disconnected from Asterisk server
>>
>> Bus error (core dumped)
>>
>> ...<snip>
>>
>>
>> I have been playing with "Local" channels over the weekend, and as cool
>> as they sound, they have caused me nothing but pain. Once again, following
>> the console log, I notice that Followme indeed uses Local channel to make
>> these calls and returns control when the call times out etc.
>>
>> The ONLY time it gets anywhere is if I use the 'l' option with Followme
>> application.
>>
>> In that case, the call connect and I can have a conversation but the
>> minute the remote party hangs up, asterisk dumps core again.
>>
>> it may be something to do with the "after return" to handle next steps
>> but what are they supposed to be? I don't want anything to happen like go
>> to VM or anything.
>>
>> Have tried this with 10.3.0 and 10.11.1. I noticed new changes have been
>> made in v11...but this should work
>>
>> How does this work?? Do I need fancy options with the "Dial" command
>> doing GoSub and what not? and Why does it insist on playing all these
>> prompts I have commented them all out from followme.conf, but it's still
>> looking to play them
>>
>> Thanks in advance
>> \A
>>
>>
>> --
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>
>
>
> --
>
> Steve Murphy
>
> ParseTree Corporation
>
> 57 Lane 17
>
> Cody, WY 82414
>
> ✉  murf at parsetree.com
>
> ☎ 307-899-5535
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
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>    http://lists.digium.com/mailman/listinfo/asterisk-users
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