[asterisk-users] Dealing with progress codes

Thu Oct 30 09:14:15 CDT 2008

Thanks for the reply!

Generally that's what I do, script local area codes and prefixes so that
dialing 1 is necessary only for long distance calls. The problem here is
that there are over 1500 area codes and prefixes (DC area) that are required
by the carrier to not be dialed with a 1 (ie, local calls). If push comes to
shove, I'll implement them all in the dialplan, but this just seems like a
poor way of handling it.

Running another Dial statement after a timeout with r will put the possible
wait time close to a minute if the callee doesn't answer and voicemail picks
up (I try to estimate as long as 30 seconds for voicemail pickup). I may
turn this on for a temporary fix, but it's not an acceptable solution long
term.

On Thu, Oct 30, 2008 at 12:08 AM, Juan Rodríguez <jerdguez at gmail.com> wrote:

> Arka:
> I thought you would reroute the call with (or without) the leading one, so,
> just Dial again.
>
> This will work and your users wont notice a BIG difference if the call is
> answered. The problem is if the call is not answer, because if you have a
> busy number, then your users will get something like "ring, ring...ring,
> beep,beep...".
>
> For a better solution I would recommend you to get at least your local
> prefixes and use the correct dial string with patterns. This can be achieved
> with a script.
>
>
> On Wed, Oct 29, 2008 at 6:15 PM, arkda <thrills3k3r at gmail.com> wrote:
>
>> I left something out on that last message, sorry.
>>
>> With r, not R, it will mask the message with ringing. I could then fail it
>> over to another dial out, however from testing I've found that my users
>> expect something to happen within 30 seconds (voicemail, pickup, etc.) The
>> worse-case scenario would be using r a time of 60 seconds. I've been
>> thinking of implementing this as a temp fix, but not something I want to
>> leave in place.
>>
>>
>>
>> On Wed, Oct 29, 2008 at 5:46 PM, arkda <thrills3k3r at gmail.com> wrote:
>>
>>> Thanks for the reply!
>>>
>>> I've played around with R to solve this (probably should have mentioned
>>> that), however I wasn't able to make it work. The message is still played
>>> (this message is from the provider). It will move to the next line in the
>>> dialplan, but as soon as users hear the message they hang up.
>>>
>>> Since the progress code comes before actual audio is played to the caller
>>> there has to be a way of catching this and dealing with it in the dialplan,
>>> but nothing I've tried so far works.
>>>
>>>
>>> On Wed, Oct 29, 2008 at 12:25 AM, Juan Rodríguez <jerdguez at gmail.com>wrote:
>>>
>>>> Try using a R or r on the Dial command, the R option is better for you
>>>> in my opinion.
>>>> i.e Dial(Zap/G2/1${EXTEN},30,R) or Dial(Zap/G2/1${EXTEN}|30|R)
>>>>
>>>> The R option is going to generate a ring tone when the callee indicates
>>>> ringing and is going wait for an Answer. As Progress is just for early
>>>> media, you wont get that message.
>>>>
>>>> For more info on the Dial command see:
>>>>
>>>> http://www.voip-info.org/wiki-Asterisk+cmd+Dial
>>>>
>>>>
>>>>
>>>> On Tue, Oct 28, 2008 at 6:56 PM, arkda <thrills3k3r at gmail.com> wrote:
>>>>
>>>>> 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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>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Juan E. Rodríguez
>>>> Cel. 829-886-5565
>>>> Work: 809-724-9227
>>>>
>>>> _______________________________________________
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>>>
>>>
>>
>> _______________________________________________
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>
>
>
> --
> Juan E. Rodríguez
> Cel. 829-886-5565
> Work: 809-724-9227
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
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