[asterisk-users] Recording with MixMonitor and AGI

Thu Mar 14 04:48:41 CDT 2013

hi,

the "music" heard by MoH is configurable... so if you want silence...
But "hold" could e.g. also be done by transferring a caller into a 
dynamic meetme room...

yves

Am 14.03.2013 08:43, schrieb Henrik Westerberg:
> Hi,
>
> The idea was to record an ongoing call by three party bridging on the 
> mobile phone.
> Well my problem was to halt execution of the Dialplan so the server 
> would not hang up the call. And I don´t want the server to say 
> anything during the call.
> Now I solved this case as well by using Answer and then Record in the 
> dialplan . So I´m not recording with MixMonitor.
>
> But just out of curiosity. How did you mean using hold (in 
> answer/hold). Is that MusicOnHold? For me I can´t use that since I 
> don´t want to make any noise. Is there another way?
>
> exten => 111,1,Answer()
> exten => 111,n,?????
>
> I have tried using Wait with a long duration but have not succeeded to 
> make it work as I want.
>
> I am using asterisk-java and originate calls to local channels.
>
> Regards,
> Henrik
>
>
> Från: "Yves A." <yves030 at gmx.de <mailto:yves030 at gmx.de>>
> Datum: söndag 10 mars 2013 11:42
> Till: Asterisk Users Mailing List - Non-Commercial Discussion 
> <asterisk-users at lists.digium.com 
> <mailto:asterisk-users at lists.digium.com>>, Henrik Westerberg 
> <henrik.westerberg at ain.se <mailto:henrik.westerberg at ain.se>>
> Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI
>
> Hi,
>
> so if your are ok with the way you solved part 1... alright, lets go 
> to part 2..
> but again... hu.. I don´t understand..
> what do you mean with merging to a mobile phone?
> do you want do bridge the calls (three partys) or do you want to play 
> the just recorded file
> from your server-initiated call into a another running call?
> what is "by hand"?
> the more explicit you are, the more helpful will be the answer.
>
> you ask "but if there is a way to just
> dial out and then let the server side of the call "Keep the channel up 
> but do nothing forever until the call is hang up""
>
> of course you can...you could e.g.:
> call into a queue
> call into a meetme room
> call with the help of a local channel into a context where you do 
> nothing but answer / hold
>
> but as i said.... i did not quite catch what your objective really 
> is... i just dont understand
> your scenario or cant imagine its sense.
>
> if you are a java programmer, i think your using the asterisk-java lib 
> from s. reuter..
> if so, you have any freedom, you could also use ami connection to 
> listen to events
> to start and stop recordings and so on.
>
> regards,
> yves
>
> Am 09.03.2013 21:32, schrieb Henrik Westerberg:
>> Hi,
>>
>> Thanks for your answer!
>>
>> 1.
>> > so you want to establish a call (triggered by ami) between two partys, record the 
>> conversation
>> > and save the file to a(nother) server (afterwards), right?
>>
>> Yes this is correct, and I prefer to do the transferring of the file 
>> to another server with my existing AGI.
>> My AGIs are written in java. Today I the upload is done over http.
>> Today I schedule the upload in the AGI script a couple of seconds 
>> after the channel is hang up. But the two
>> lines might not be hung up at the same time.
>> Your suggestion of always fixing the file is wise, it now seems to 
>> work fine after having been processed with sox.
>>
>> So now I think that this case 1 is ok for me :-)
>>
>> 2.
>> > and another task is to establish (also ami triggered) a call to a mobile and play, 
>> lets say a voicefile.
>> > this "conversation" should also be recorded and saved on a(nother) 
>> server (afterwards), right?
>>
>> The idea is to perform a "probe call" with the only task of recording 
>> what the other party says.
>> It will be merged "by hand" on a mobile phone to an ongoing call with 
>> another party.
>> This could be done by calling out and letting AGI execute a RECORD 
>> FILE but if there is a way to just
>> dial out and then let the server side of the call "Keep the channel 
>> up but do nothing forever until the call is hang up"
>> Then I could easily use the MixMonitor and write the whole 
>> conversation in the dialplan with uploading similar to the first case.
>> Any suggestions?
>>
>> Regards,
>> Henrik
>>
>>
>>
>> Från: "Yves A." <yves030 at gmx.de <mailto:yves030 at gmx.de>>
>> Svara till: Asterisk Users Mailing List - Non-Commercial Discussion 
>> <asterisk-users at lists.digium.com 
>> <mailto:asterisk-users at lists.digium.com>>
>> Datum: torsdag 7 mars 2013 20:10
>> Till: Asterisk Users Mailing List - Non-Commercial Discussion 
>> <asterisk-users at lists.digium.com 
>> <mailto:asterisk-users at lists.digium.com>>
>> Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI
>>
>> hi,
>>
>> hard to understand, what your objective is... at least for me ;-)
>>
>> so you want to establish a call (triggered by ami) between two 
>> partys, record the conversation
>> and save the file to a(nother) server (afterwards), right?
>>
>> and another task is to establish (also ami triggered) a call to a 
>> mobile and play, lets say a voicefile.
>> this "conversation" should also be recorded and saved on a(nother) 
>> server (afterwards), right?
>>
>> let me know, if i understood you right, the solution is not so hard 
>> to implement.
>> In what language do you preferrably write your AGIs? (although there 
>> is no absolute need for using an
>> agi... you can all write down in your dialplan...)
>> is there a special protocol requirement for saving/transferring the 
>> recorded voicefile (e.g. ftps)?
>> One obstacle is, that the recorded file is not fully written 
>> _immediately_ after stopmixmonitor or hangup...
>> this has to be taken care of and depending on your agi... it might be 
>> interrupted, if the call is hungup...
>> but as you did not show your agi... these are just hints..
>>
>> regards,
>> yves
>>
>>
>>
>> Am 07.03.2013 16:21, schrieb Henrik Westerberg:
>>> Hi,
>>>
>>> I am developing a call recording application on Asterisk 11.2 and 
>>> have this configuration in my dialplan:
>>>
>>> [macro-ccdev2-rec]
>>> exten => s,1,MixMonitor(${ARG1},b)
>>>
>>> [outgoing-originate]
>>> exten => _X.,1,NoOp(Will send call to ${EXTEN})
>>> exten => _X.,n,Dial(SIP/${EXTEN}@x.y.z)
>>>
>>> [outgoing-originate-rec]
>>> exten => 
>>> h,1,Agi(agi://localhost/ajpbx.agi?path=uploadrec&callid=${CC_CALLID})
>>>
>>> exten => _X,1,NoOp(Will send call to ${EXTEN}, CC_CALLID is 
>>> ${CC_CALLID}, CC_FILENAME is ${CC_FILENAME})
>>> exten => _X,n,Dial(SIP/${EXTEN}@x.y.z,60,M(ccdev2-rec^${CC_FILENAME})e)
>>>
>>> If I want to make a recorded server callout from 077777777 
>>> to 0888888888 I then originate a call via AMI to 
>>> Local/077777777 at outgoing-originate with context set 
>>> to outgoing-originate-rec and extension to 0888888888.
>>> The result will be something like this:
>>>
>>>     -- Executing [s at macro-ccdev2-rec:1] 
>>> MixMonitor("SIP/upps-ccm-tq01-0000003f", "cbrec-15605.wav,b") in new 
>>> stack
>>>   == Begin MixMonitor Recording SIP/upps-ccm-tq01-0000003f
>>>     -- Executing [h at outgoing-originate-rec:1] 
>>> AGI("SIP/upps-ccm-tq01-0000003e", 
>>> "agi://l4574/ajpbxtest.agi?path=uploadrec&callid=15605") in new stack
>>>     -- <SIP/upps-ccm-tq01-0000003e>AGI Script 
>>> agi://localhost/ajpbxtest.agi?path=uploadrec&callid=15605 completed, 
>>> returning 0
>>>     -- Executing [h at outgoing-originate-rec-dev2:1] 
>>> AGI("SIP/upps-ccm-tq01-0000003f", 
>>> "agi://4574/ajpbxtest.agi?path=uploadrec&callid=") in new stack
>>>     -- <SIP/upps-ccm-tq01-0000003f>AGI Script 
>>> agi://localhost/ajpbxtest.agi?path=uploadrec&callid= completed, 
>>> returning 0
>>>   == MixMonitor close filestream (mixed)
>>>   == End MixMonitor Recording SIP/upps-ccm-tq01-0000003f
>>>
>>> Unfortunately I get two different calls to the h extension, but this 
>>> I can cope with. The one without called is not interesting.
>>> The uploading will fail since the MixMonitor is still on when I try 
>>> to upload the file. The file will not have a duration. It works when 
>>> I schedule the uploading a while after from my agi application but I 
>>> would rather not rely on a timeout.
>>>
>>> When I tried to run StopMixMonitor before the Agi call in the h 
>>> extension, the first call fail and I never get any uploading with 
>>> callid.
>>>
>>>     -- Executing [s at macro-ccdev2-rec:1] 
>>> MixMonitor("SIP/upps-ccm-tq01-00000043", "cbrec-15607.wav,b") in new 
>>> stack
>>>   == Begin MixMonitor Recording SIP/upps-ccm-tq01-00000043
>>>     -- Executing [h at outgoing-originate-rec-dev2:1] 
>>> StopMixMonitor("SIP/upps-ccm-tq01-00000042", "") in new stack
>>>   == Spawn extension (outgoing-originate-rec-dev2, h, 1) exited 
>>> non-zero on 'SIP/upps-ccm-tq01-00000042'
>>>     -- Executing [h at outgoing-originate-rec-dev2:1] 
>>> StopMixMonitor("SIP/upps-ccm-tq01-00000043", "") in new stack
>>>   == MixMonitor close filestream (mixed)
>>>     -- Executing [h at outgoing-originate-rec-dev2:2] 
>>> AGI("SIP/upps-ccm-tq01-00000043", 
>>> "agi://localhost/ajpbxtest.agi?path=uploadrec&callid=") in new stack
>>>
>>> Am I missing something here? I also looked at the possibility to 
>>> specify a command to execute when MixMonitor stops but I would 
>>> rather handle the file uploading in my agi application.
>>>
>>> I also have another case: I want to dial out a call and record it. 
>>> It will be a "oneway-call" from the server to a mobile. Do I need to 
>>> get AGI-control of it and record with an AGI command or how can I 
>>> hack it directly in the dial plan using MixMonitor?
>>>
>>> Best Regards,
>>> Henrik
>>>
>>>
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>>
>>
>>
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>
>
>
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