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 <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]>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 <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> 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 [EMAIL PROTECTED]:1] Set("SIP/user9-b696fb58",
>>>>> "GROUP(default)=dialpool") in new stack
>>>>> -- Executing [EMAIL PROTECTED]:2] GotoIf("SIP/user9-b696fb58",
>>>>> "1?5") in new stack
>>>>> -- Goto (internal,5551515121,5)
>>>>> -- Executing [EMAIL PROTECTED]:5] Set("SIP/user9-b696fb58",
>>>>> "GROUP(default)=dialpool") in new stack
>>>>> -- Executing [EMAIL PROTECTED]:6] Answer("SIP/user9-b696fb58",
>>>>> "") in new stack
>>>>> -- Executing [EMAIL PROTECTED]:7] Set("SIP/user9-b696fb58",
>>>>> "CALLERID(num)=5552223333") in new stack
>>>>> -- Executing [EMAIL PROTECTED]:8] Set("SIP/user9-b696fb58",
>>>>> "CALLERID(name)=HiThere") in new stack
>>>>> -- Executing [EMAIL PROTECTED]:9] NoOp("SIP/user9-b696fb58",
>>>>> "--out the pri--") in new stack
>>>>> -- Executing [EMAIL PROTECTED]: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($[${GROUP_COUNT([EMAIL PROTECTED])}<19]?5)
>>>>> exten => _NXXXXXXXXX,3,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED]
>>>>> )}>18]?BLOCK)
>>>>> exten => _NXXXXXXXXX,4,NoOp
>>>>> exten => _NXXXXXXXXX,5,Set(GROUP(default)=dialpool)
>>>>> exten => _NXXXXXXXXX,6,Answer()
>>>>> exten => _NXXXXXXXXX,7,Set(CALLERID(num)=${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
>>>
>>> _______________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
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>>>
>>
>>
>
> _______________________________________________
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> asterisk-users mailing list
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--
Juan E. Rodríguez
Cel. 829-886-5565
Work: 809-724-9227
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