[asterisk-users] Dealing with progress codes

[arkda]

Some additional information.

I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an unusual
result:

[Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical Response)

This occurs about a second after the user hangs up on the error message
being played from the provider. I have a feeling it's trying to execute the
next step in the dialplan but unable since the caller hung up.

Thoughts, criticism, insults all welcome!

On Tue, Oct 28, 2008 at 12:53 PM, arkda <thrills3k3r at gmail.com> wrote:

> Hi,
>
> I've ran into an issue with a PRI provider in a major metropolitan area
> that I haven't needed to deal with before and I was hoping someone might
> have some insight on how to handle this within the Asterisk dialplan.
>
> At this location users can't always tell if a number is long distance or
> not (there are a lot of area codes and prefixes in the vicinity).
> Additionally, users are required by the provider to dial the full 10 digit
> number even if a call is local since a local call could be for a few
> different area codes and prefixes. The problem is the provider requires a 1
> in front of the number for long distance calls, but errors out if the call
> has a 1 in front and the call is local.
>
> As a result, users are complaining that they are constantly having to
> redial with or without the 1. I've tracked down this behavior when a call
> fails:
>
>     -- Executing [5551515121 at internal:1] Set("SIP/user9-b696fb58",
> "GROUP(default)=dialpool") in new stack
>     -- Executing [5551515121 at internal:2] GotoIf("SIP/user9-b696fb58",
> "1?5") in new stack
>     -- Goto (internal,5551515121,5)
>     -- Executing [5551515121 at internal:5] Set("SIP/user9-b696fb58",
> "GROUP(default)=dialpool") in new stack
>     -- Executing [5551515121 at internal:6] Answer("SIP/user9-b696fb58", "")
> in new stack
>     -- Executing [5551515121 at internal:7] Set("SIP/user9-b696fb58",
> "CALLERID(num)=5552223333") in new stack
>     -- Executing [5551515121 at internal:8] Set("SIP/user9-b696fb58",
> "CALLERID(name)=HiThere") in new stack
>     -- Executing [5551515121 at internal:9] NoOp("SIP/user9-b696fb58", "--out
> the pri--") in new stack
>     -- Executing [5551515121 at internal:10] Dial("SIP/user9-b696fb58",
> "Zap/G2/15551515121") in new stack
>     -- Requested transfer capability: 0x00 - SPEECH
>     -- Called G2/15551515121
>     -- Zap/22-1 is proceeding passing it to SIP/user9-b696fb58
>     -- PROGRESS with cause code 31 received
>     -- Zap/22-1 is making progress passing it to SIP/user9-b696fb58
>     -- Hungup 'Zap/22-1'
>   == Spawn extension (internal, 5551515121, 10) exited non-zero on
> 'SIP/user9-b696fb58'
>
> The above call was a call that is considered local by the provider. The
> caller is then redirected to a message (by the provider) saying 'You do not
> need to dial a one or zero...' and the message repeats indefinitely.
>
> I'd like to figure out how to handle this in the dial plan so users do not
> even know anything happened. To test to see if I could stop the call
> progress and reroute it I've tried this so far:
>
> exten => _NXXXXXXXXX,1,Set(GROUP(default)=dialpool)
> exten => _NXXXXXXXXX,2,GotoIf($[${GROUP_COUNT(priout at default)}<19]?5)
> exten => _NXXXXXXXXX,3,GotoIf($[${GROUP_COUNT(congest at default)}>18]?BLOCK)
> exten => _NXXXXXXXXX,4,NoOp
> exten => _NXXXXXXXXX,5,Set(GROUP(default)=dialpool)
> exten => _NXXXXXXXXX,6,Answer()
> exten => _NXXXXXXXXX,7,Set(CALLERID(num)=${CLR})
> exten => _NXXXXXXXXX,8,Set(CALLERID(name)=HiThere)
> exten => _NXXXXXXXXX,9,NoOp(--out the pri--)
> ; Primary Dialout
> exten => _NXXXXXXXXX,10,Dial(Zap/G2/1${EXTEN})
> exten => _NXXXXXXXXX,11,GotoIf,($[${HANGUPCAUSE} = 31]?YAY)
> exten => _NXXXXXXXXX,12,Hangup()
> ; Call limiter
> exten => _NXXXXXXXXX,n(BLOCK),Answer()
> exten => _NXXXXXXXXX,n(BLOCK),Playback(all-circuits-busy-now)
> exten => _NXXXXXXXXX,n(BLOCK),Playback(pls-try-call-later)
> exten => _NXXXXXXXXX,n(BLOCK),Hangup()
> ; 1 tester
> exten => _NXXXXXXXXX,n(YAY),Answer()
> exten => _NXXXXXXXXX,n(YAY),Playback(beep)
> exten => _NXXXXXXXXX,n(YAY),Hangup()
>
> It doesn't work. The user simply hangs up when the message is heard and the
> next line in the dialplan isn't followed. How can I detect that a call has
> received a progress code 31 then reroute it to another extension? From what
> I found on voip-info.org (
> http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+hangupcause) this should work. What am I missing?
>
> The server is running Asterisk 1.4.21.1, zaptel 1.4.11, libpri 1.4.5,
> compiled from source.
>
> Thanks in advance!
>
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