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!
>>>>
>>>
>>>
>>> _______________________________________________
>>> -- 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
>>>
>>
>>
>>
>> --
>> 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
>>
>
>
_______________________________________________
-- 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
