[Asterisk-Users] Dial + disconnect
Richard Lyman
pchammer at dynx.net
Wed Sep 10 22:19:30 MST 2003
also, when 'spooling' calls you are originating a call, then IF connect
tossing it back into the extension processing (if you use an exten (pbx
based event). so, a busy would be for that call, not like an outgoing
context that would use say... congestion
please, download and read. www.digium.com/handbook-draft.pdf
Richard Lyman wrote:
> well getting back to what hardware and WHERE ...
> you must not have looked at the zapata.conf (which is where i
> suggested you look in the first reply)
> note the last line of the documentation
> hmm...
>
> ; On trunk interfaces (FXS) it can be useful to attempt to follow the
> progress
> ; of a call through RINGING, BUSY, and ANSWERING. If turned on, call
> ; progress attempts to determine answer, busy, and ringing on phone
> lines.
> ; This feature is HIGHLY EXPERIMENTAL and can easily detect false
> answers,
> ; so don't count on it being very accurate. Also, it is ONLY
> configured for
> ; standard U.S. tones
> ;
> callprogress=yes
>
>
> Chee Foong wrote:
>
>> Hi,
>>
>> Sorry, forgot about the info. I have a E100P digium card and is
>> planing to
>> implement an automatic dialing system using the spooling mechanism.
>> In this
>> service, asterisk will call Party A then Party B and finally let them
>> talk.
>> As you know, Party A or Party B might not be reached. Therefore I
>> need to
>> know, if a particular party is busy, the party does not answer the
>> call or
>> the call get through well. This is to determine whether to retry the
>> call if
>> something fail or do some other operations.
>>
>> I looked at the handbooks, tutorial on digium.com and also the
>> mailing list
>> but nothing really related. I checked the dial application through
>> the CLI,
>> It mention something about the priority n+101 if dial return busy,
>> but if I
>> am not mistaken that is only happend if no channel is available (in
>> my case
>> 30 channels has been occupied). It is not the destination party that is
>> busy.
>> Am I right?
>>
>> Thanks for you reply
>>
>>
>> ----- Original Message -----
>> From: "Richard Lyman" <pchammer at dynx.net>
>> To: <asterisk-users at lists.digium.com>
>> Sent: Thursday, September 11, 2003 12:03 PM
>> Subject: Re: [Asterisk-Users] Dial + disconnect
>>
>>
>>
>>
>>> well, you never said what hardware you had, and where you were using
>>> it... so
>>>
>>> even if everything is working... look at handbook for extension
>>> handling. you'll also have to work out the mysql updates, given that
>>> the only current sql module is app_sql-postresql (or there abouts).
>>> (there used to be some mysql (mostly for cdr) floating about the
>>> mailing
>>> list (search the archives)).
>>>
>>> www.digium.com/handbook-draft.pdf
>>> has alot of helpful info, you should take a peek.
>>>
>>> Chee Foong wrote:
>>>
>>>
>>>
>>>> Ahhh...I see those config options.
>>>>
>>>> But how would I know which case(busy, no answer) happen in the
>>>> extension.conf.
>>>> What I am planing to do is somthing like:
>>>> if(Busy detected)
>>>> put a retry flag in MySQL db, for retying later
>>>> else if(no answer detected)
>>>> then stop trying the destination number, and log the call info in
>>>>
>>>
>> MySQL.
>>
>>
>>>> thanks
>>>>
>>>> Foong
>>>>
>>>>
>>>> ----- Original Message -----
>>>> From: "Richard Lyman" <pchammer at dynx.net>
>>>> To: <asterisk-users at lists.digium.com>
>>>> Sent: Thursday, September 11, 2003 12:18 AM
>>>> Subject: Re: [Asterisk-Users] Dial + disconnect
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>> well depending on the hardware you are using and where you are
>>>>> using it at, in some cases there is. look in zapata.conf search
>>>>> 'callprogress'.
>>>>>
>>>>> Chee Foong wrote:
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>> Yes you are right, Sorry my mistake.
>>>>>>
>>>>>> So, is there a way to detect busy, answer, or no answer call?
>>>>>>
>>>>>> Foong
>>>>>>
>>>>>> ----- Original Message -----
>>>>>> From: "Richard Lyman" <pchammer at dynx.net>
>>>>>> To: <asterisk-users at lists.digium.com>
>>>>>> Sent: Wednesday, September 10, 2003 12:50 PM
>>>>>> Subject: Re: [Asterisk-Users] Dial + disconnect
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>> based on what dial string you have a zap device '0122740900' (looks
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>> more
>>>>
>>>>
>>>>
>>>>
>>>>>>> like an exten/phone# to me) maybe you meant Zap/1/0122740900
>>>>>>> (notice
>>>>>>> the /1/, you could also use groups /g1/ (if setup in zapata.conf))
>>>>>>>
>>>>>>> Chee Foong wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> Hello,
>>>>>>>>
>>>>>>>> When I have the following extension:
>>>>>>>>
>>>>>>>> exten => 900,1,dial(Zap/0122740900)
>>>>>>>>
>>>>>>>> can I know whether 'dial' actually gets through or the called
>>>>>>>> party
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>> is
>>>>
>>>>
>>>>
>>>>
>>>>>>>> busy at the moment. I want to perform different action based on
>>>>>>>> whether the 'dail' success or not.
>>>>>>>>
>>>>>>>>
>>>>>>>> Foong
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Asterisk-Users mailing list
>>>>>>> Asterisk-Users at lists.digium.com
>>>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
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>>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
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>>>>>
>>>>>
>>>>>
>>>>>
>>>>
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>>>>
>>>>
>>>>
>>>
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>>
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>
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