asterisk users - Apr 2005 - how to tell what ${DIALSTATUS} is being set

i often have nufone problems, e.g.

    -- Executing Dial("SIP/konaa0p-4b88",
"IAX2/myname@NuFone/14086661234") in new stack
    -- Called myname@NuFone/14086661234
    -- Call accepted by 66.225.202.72 (format ulaw)
    -- Format for call is ulaw
    -- Hungup 'IAX2/NuFone/5'

<sound of surf (on a boogie board kind of day) for a fairly long while>

  == No one is available to answer at this time
    -- Executing Hangup("SIP/konaa0p-4b88", "") in new stack
  == Spawn extension (dial-gateways, 14086661234, 5) exited non-zero on
'SIP/konaa0p-4b88'
    -- Executing Hangup("SIP/konaa0p-4b88", "") in new stack
  == Spawn extension (dial-gateways, h, 1) exited non-zero on
'SIP/konaa0p-4b88'

i would like to detect this (and many other things) in ${DIALSTATUS}
conditions so that i can then GotoIf() them.  the problem is that the
log does not tell me explicitly which ${DIALSTATUS} has been returned,
leaving me guessing.  with BUSY vs CONGESTION this is even more of an
issue.

is it reasonable to ask that the log contain the value being set in
${DIALSTATUS}?

randy

Have you tried a simple hangup NoOp to output ${DIALSTATUS} to the CLI?

exten => h,1,NoOp("${DIALSTATUS}")

-josiah


Original Message:
-----------------
From: Randy Bush randy@psg.com
Date: Sat, 2 Apr 2005 10:44:46 -0800
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] how to tell what ${DIALSTATUS} is being set


i often have nufone problems, e.g.

    -- Executing Dial("SIP/konaa0p-4b88",
"IAX2/myname@NuFone/14086661234")
in new stack
    -- Called myname@NuFone/14086661234
    -- Call accepted by 66.225.202.72 (format ulaw)
    -- Format for call is ulaw
    -- Hungup 'IAX2/NuFone/5'

<sound of surf (on a boogie board kind of day) for a fairly long while>

  == No one is available to answer at this time
    -- Executing Hangup("SIP/konaa0p-4b88", "") in new stack
  == Spawn extension (dial-gateways, 14086661234, 5) exited non-zero on
'SIP/konaa0p-4b88'
    -- Executing Hangup("SIP/konaa0p-4b88", "") in new stack
  == Spawn extension (dial-gateways, h, 1) exited non-zero on
'SIP/konaa0p-4b88'

i would like to detect this (and many other things) in ${DIALSTATUS}
conditions so that i can then GotoIf() them.  the problem is that the
log does not tell me explicitly which ${DIALSTATUS} has been returned,
leaving me guessing.  with BUSY vs CONGESTION this is even more of an
issue.

is it reasonable to ask that the log contain the value being set in
${DIALSTATUS}?

randy

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In article <16974.59422.117207.124524@roam.psg.com>,
Randy Bush <randy@psg.com> wrote:
> i often have nufone problems, e.g.
> 
>     -- Executing Dial("SIP/konaa0p-4b88",
"IAX2/myname@NuFone/14086661234") in new stack
>     -- Called myname@NuFone/14086661234
>     -- Call accepted by 66.225.202.72 (format ulaw)
>     -- Format for call is ulaw
>     -- Hungup 'IAX2/NuFone/5'
> 
> <sound of surf (on a boogie board kind of day) for a fairly long
while>
> 
>   == No one is available to answer at this time
>     -- Executing Hangup("SIP/konaa0p-4b88", "") in new
stack
>   == Spawn extension (dial-gateways, 14086661234, 5) exited non-zero on
'SIP/konaa0p-4b88'
>     -- Executing Hangup("SIP/konaa0p-4b88", "") in new
stack
>   == Spawn extension (dial-gateways, h, 1) exited non-zero on
'SIP/konaa0p-4b88'
> 
> i would like to detect this (and many other things) in ${DIALSTATUS}
> conditions so that i can then GotoIf() them.  the problem is that the
> log does not tell me explicitly which ${DIALSTATUS} has been returned,
> leaving me guessing.  with BUSY vs CONGESTION this is even more of an
> issue.
> 
> is it reasonable to ask that the log contain the value being set in
> ${DIALSTATUS}?
I find it useful to follow the Dial command with a NoOp command as follows:
 NoOp(DIALSTATUS=${DIALSTATUS})

Then it shows up in the log. It also provides something for my Manager event
parser to see, in order to discover the reason for a failed call.

Cheers
Tony
-- 
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org

Hi for all!

I saw it on http://signate.com/features.php an Open Source PBX Features with
support Cisco Skinny Call Control Protocol.

Is it possible in Asterisk or I need a license for this?

Has anyone using Asterisk with Cisco Skinny?

TIA

Alexandre