[asterisk-users] Zap Bridged Calls do not continue dialplan

Wed Jul 9 11:51:22 CDT 2008

Well, i tried to see a noop with:

exten = 205,1,Dial(Zap/g1/205)
exten = 205,2,NoOp(After Dial ${DIALSTATUS})
exten = 205,102,VoiceMail
exten = 205,103,NoOp(After VM ${DIALSTATUS})

to see i got any dial status changes, i got nothing.

Let me explain my path and mabye it might help.
A User calls in the DID, this goes to the legacy pbx.
The PBX forwards the calls accross the 4 pots lines to the asterisk box.
Asterisk answers the call and plays a menu.
[voicemenu-custom-1]
comment = AutoAttendMain
alias_exten = 299
include = default
exten = s,1,Answer
exten = s,2,Background(record/afternoon)
exten = s,3,Wait(20)
exten = s,4,Playback(goodbye)
exten = s,6,Hangup
exten = 0,1,Goto(voicemenu-custom-2|s|1)
exten = 1,1,Goto(default|380|1)
exten = 3,1,Goto(default|381|1)
exten = 4,1,Goto(default|382|1)
exten = 5,1,Goto(default|383|1)
exten = 6,1,Goto(default|384|1)
exten = 7,1,Goto(voicemenu-custom-3|s|1)
exten = *,1,Goto(default|298|1)
exten = #,1,Goto(default|777|1)

Now since it is allowed to dial other extensions from the menu, they
dial for example, 205
Asterisk then dials to 205(205 is in default, so it should go through
it's priority list right?)
But it doesnt seem to, nor do i see it set a dialstatus variable in
the debug log
here is the debug log
[Jul  9 09:43:41] DEBUG[2068] chan_zap.c: Dialing '205'
[Jul  9 09:43:41] DEBUG[2068] chan_zap.c: Deferring dialing...
[Jul  9 09:43:42] DEBUG[2068] chan_zap.c: Sent deferred digit string: T205w
[Jul  9 09:43:43] DEBUG[2068] chan_zap.c: master: 2, slave: 1, nothingok: 0
[Jul  9 09:43:43] DEBUG[2068] chan_zap.c: Stopping tones on 2/0 talking to 1/0
[Jul  9 09:43:43] DEBUG[2068] chan_zap.c: Stopping tones on 1/0 talking to 2/0
[Jul  9 09:43:43] DEBUG[2068] chan_zap.c: Making 1 slave to master 2 at 0
[Jul  9 09:43:43] DEBUG[2068] chan_zap.c: Added 23 to conference 9/2
[Jul  9 09:43:43] DEBUG[2068] chan_zap.c: Added 24 to conference 9/1
[Jul  9 09:43:45] DEBUG[2068] dsp.c: ast_dsp_busydetect detected busy,
avgtone: 245, avgsilence 220
[Jul  9 09:43:45] DEBUG[2068] chan_zap.c: Unlinking slave 1 from 2
[Jul  9 09:43:45] DEBUG[2068] chan_zap.c: Removed 23 from conference 9/2
[Jul  9 09:43:45] DEBUG[2068] chan_zap.c: Removed 24 from conference 9/1

and here is the sample from the console(set to vvvv)
-- Starting simple switch on 'Zap/2-1'
[Jul  9 09:43:37] NOTICE[2068]: chan_zap.c:6387 ss_thread: Got event
18 (Ring Begin)...
[Jul  9 09:43:38] NOTICE[2068]: chan_zap.c:6387 ss_thread: Got event 2
(Ring/Answered)...
  == Starting Zap/2-1 at DID_trunk_1,s,1 failed so falling back to exten 's'
  == Starting Zap/2-1 at DID_trunk_1,s,1 still failed so falling back
to context 'default'
    -- Executing [s at default:1] GotoIfTime("Zap/2-1",
"06:00-11:00|mon-fri|*|*?voicemenu-custom-4|s|1") in new stack
    -- Goto (voicemenu-custom-4,s,1)
    -- Executing [s at voicemenu-custom-4:1] Answer("Zap/2-1", "") in new stack
    -- Executing [s at voicemenu-custom-4:2] BackGround("Zap/2-1",
"record/Morning") in new stack
    -- <Zap/2-1> Playing 'record/Morning' (language 'en')
  == CDR updated on Zap/2-1
    -- Executing [205 at voicemenu-custom-4:1] Dial("Zap/2-1",
"Zap/g1/205") in new stack
    -- Called g1/205
    -- Zap/1-1 answered Zap/2-1
    -- Native bridging Zap/2-1 and Zap/1-1
    -- Hungup 'Zap/1-1'
  == Spawn extension (voicemenu-custom-4, 205, 1) exited non-zero on 'Zap/2-1'
    -- Hungup 'Zap/2-1'

I really appreciate all the help!
On Wed, Jul 9, 2008 at 7:16 AM, Mr Shunz <mrshunz at gmail.com> wrote:
>> Hello all!
>
> Hello!
>
>> I'm having problem with the calls that come through my asterisk box
>> and back out to our legacy pbx, it seems to be that even if the call
>> is ringing and not picked up yet, zap reports the line as answered,
>> why is it doing that?
>
> could be that the PBX *answers* the line (maybe for playing moh/messages)
> and then rings the phones connected...
>
>> Also it wont stay connected for the correct amount of rings, wont
>> continue the dialplan after zap makes it hangup. The same thing
>> happens if the line is busy from dialing the legacy pbx.
>
> I think that if it hangup asterisk (by default) jumps to extension h,
> so you may want create smth like
>
>    exten => h,1,...
>
> if you'd like to do other things after hangups
>
> [snip]
>
>> it seems to have to do with the busydetect=yes
>> it detects it but just hangs up.
>>
>> Is there a way to tell it to just stop the call and continue with the dialplan?
>
> you can check the ${DIALSTATUS} var and jump accordingly,
> like:
>
>    exten => s,n,Goto(s-${DIALSTATUS},1)
>
> and then have
>
>    exten => s-BUSY,1,...
>
>    exten => s-NOANSWER,1,...
>
> and so on ...
>
>> Some more info i have theorized, it seems that since the other channel
>> "answers" even though it is dialing out and only ringing(or busy or
>> whatever), it thinks that the call was picked up.
>
> yeah, that's it...
>
>> I might be able to work around this with call confirmation. i'm gonna
>> try that today, is there any way to do a call confirmation by just
>> detecting if there is a voice on the other end?
>> Also, the calls even while normal ringing time out to fast, i cant
>> seem to find that setting, where might it be globally?
>
> don't know about these ...
>
> cheers
>
> --
> ------------------------------------------------
> Daniele Santi .o.
> daniele at santi.vr.it ..o () ascii ribbon campaign
> Linux User #415108 ooo /\ www.asciiribbon.org
> ------------------------------------------------
>
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