[asterisk-users] Calls dropping

[John Regal]

Hi,
I am using Originate in testing and also using call files in testing. I also
needed to capture DIALSTATUS and update my CDRs accordingly. My original
attempts at using Originate (or call files) did not report {DIALSTATUS} if
the call could not be connected (e.g. bad phone number like 555-555-5555)
because, it seems, the call never entered the dialplan context defined in
Originate. So, my latest code uses Local/mycontext/n

So it appears, intermittently, the channel disappears (fails?). This is the
output of an instance of failure. I have also listed the code from my
context and the AMI call. A more readable version is attached. I am 1.6.1.1
- Thanks for looking...


    -- Executing [dialnumber at dialthem_private:2]
Dial("Local/dialnumber at dialthem_private-c64a;2",
"SIP/12122221111 at flowroute,30,ghM(dialthem_private^callJohnSmith^SIP/1212222
1111 at flowroute)") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 12122221111 at flowroute


    -- SIP/flowroute-0821ffc8 is making progress passing it to
Local/dialnumber at dialthem_private-c64a;2
  == Spawn extension (dialthem_private, dialnumber, 2) exited non-zero on
'Local/dialnumber at dialthem_private-c64a;2'
[Jun 30 19:52:01] ERROR[26664]: pbx.c:8637 device_state_cb: Received invalid
event that had no device IE
[Jun 30 19:52:01] ERROR[26664]: app_queue.c:810 device_state_cb: Received
invalid event that had no device IE

[dialthem_private]
exten => dialnumber,1,UserEvent(BeforeDial,ActionID:${INSP_ActionID} &
${UNIQUEID} & ${CHANNEL} & ${INSP_DialInfo} & ${INSP_$
exten =>
dialnumber,n,Dial(${INSP_DialInfo},${INSP_RingTimeout},ghM(INSP_private^${IN
SP_ActionID}^${INSP_DialInfo}))
exten => dialnumber,n,UserEvent(AfterDial,ActionID:${INSP_ActionID} &
${UNIQUEID} & ${CHANNEL} & ${INSP_DialInfo} & ${DIALST$
exten => dialnumber,n,Hangup()

[macro-INSP_private]
exten => s,1,UserEvent(SIPDial,ActionID:${ARG1} & ${UNIQUEID} & ${CHANNEL} &
${ARG2})


http://192.168.1.2:8088/asterisk/rawman?action=Originate&channel=Local%2Fdia
lnumber%40dialthem_private%2Fn&exten=s&context=detect&priority=1&CallerID=19
999999999&async=1&actionID=callingJohnSmith&account=myaccountvalue&variable=
INSP_ActionID%3DcallJohn&variable=INSP_DialInfo%3DSIP%2F12122221111%40flowro
ute&variable=INSP_RingTimeout%3D30&variable=phonenumber%3D2122221111&variabl
e=file%3DFA3469AC-BCDE-E6EB-B3AA936266704744&variable=alertID%3DFA3469AC-BCD
E-E6EB-B3AA936266704744&variable=subscriberID%3D1234512345

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David
Backeberg
Sent: Friday, June 26, 2009 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls dropping

On Thu, Jun 25, 2009 at 9:55 PM, John Regal<jregal at gmail.com> wrote:
> When using this method, it appears that the call file creates the first
part
> of the call, then creates a second call with the Dial() app. Once the call
> executed by the Dial() app is answered, the two calls are joined together.
> What I am experiencing is that sometime the first part of the call drops
and
> therefore is never joined to the second part of the call. I see errors
like

I don't quite understand what you're trying to do, but it sounds like
call two parties and join them together. Perhaps you'd prefer to use
Originate() via AMI rather than the dialplan and extension approach
you're using now?

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