Re: [asterisk-users] Recording with MixMonitor and AGI
Hi, Ok, thanks. /Henrik Från: Yves A. yves...@gmx.demailto:yves...@gmx.de Svara till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Datum: torsdag 14 mars 2013 10:48 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI 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. yves...@gmx.demailto:yves...@gmx.de Datum: söndag 10 mars 2013 11:42 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com, Henrik Westerberg henrik.westerb...@ain.semailto:henrik.westerb...@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. yves...@gmx.demailto:yves...@gmx.de Svara till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Datum: torsdag 7 mars 2013 20:10 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@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
Re: [asterisk-users] Recording with MixMonitor and AGI
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. yves...@gmx.demailto:yves...@gmx.de Datum: söndag 10 mars 2013 11:42 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com, Henrik Westerberg henrik.westerb...@ain.semailto:henrik.westerb...@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. yves...@gmx.demailto:yves...@gmx.de Svara till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Datum: torsdag 7 mars 2013 20:10 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@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
Re: [asterisk-users] Recording with MixMonitor and AGI
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. yves...@gmx.de mailto:yves...@gmx.de Datum: söndag 10 mars 2013 11:42 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com, Henrik Westerberg henrik.westerb...@ain.se mailto:henrik.westerb...@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. yves...@gmx.de mailto:yves...@gmx.de Svara till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Datum: torsdag 7 mars 2013 20:10 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@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
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. yves...@gmx.de mailto:yves...@gmx.de
Svara till: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com
Datum: torsdag 7 mars 2013 20:10
Till: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com mailto:asterisk-users@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=uploadreccallid=${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 0
to 08 I then originate a call via AMI to
Local/0@outgoing-originate with context set
to outgoing-originate-rec and extension to 08.
The result will be something like
Re: [asterisk-users] Recording with MixMonitor and AGI
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. yves...@gmx.demailto:yves...@gmx.de
Svara till: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com
Datum: torsdag 7 mars 2013 20:10
Till: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.commailto:asterisk-users@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.zmailto:EXTEN}@x.y.z)
[outgoing-originate-rec]
exten = h,1,Agi(agi://localhost/ajpbx.agi?path=uploadreccallid=${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,Mmailto:EXTEN}@x.y.z,60,M(ccdev2-rec^${CC_FILENAME})e)
If I want to make a recorded server callout from 0 to 08 I then
originate a call via AMI to Local/0@outgoing-originate with context set
to outgoing-originate-rec and extension to 08.
The result will be something like this:
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-003f, cbrec-15605.wav,b) in new stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-003f
-- Executing [h@outgoing-originate-rec:1] AGI(SIP/upps-ccm-tq01-003e,
agi://l4574/ajpbxtest.agi?path=uploadreccallid=15605) in new stack
-- SIP/upps-ccm-tq01-003eAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid=15605 completed, returning 0
-- Executing [h@outgoing-originate-rec-dev2:1]
AGI(SIP/upps-ccm-tq01-003f,
agi://4574/ajpbxtest.agi?path=uploadreccallid=) in new stack
-- SIP/upps-ccm-tq01-003fAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid= completed, returning 0
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/upps-ccm-tq01-003f
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
Re: [asterisk-users] Recording with MixMonitor and AGI
Hi,
Ok but when I use the macro the recording doesn´t start until the call is
answered which is a plus. It´s easy to trim away silence of course though.
But according to the documentation it seems like DeadAgi is obsolete in
Asterisk 1.6 and later, that AGI should be used instead.
Regards,
Henrik
Den 2013-03-08 05:30 skrev Bharat Lalcheta bharatlalch...@gmail.com:
As far as i understand your requirements, there is no need to use
macro for recording, You can directly call mixmonitor before Dial
application in your dialplan with required options. For transfer of
file, you are using AGI in h priority. However, you have to use
DeadAgi in h extension. As your channel already hangup, it can not
run on AGI.
Hope it will help you.
Regards,
Bharat Lalcheta
On Thu, Mar 7, 2013 at 8:51 PM, Henrik Westerberg
henrik.westerb...@ain.se wrote:
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=uploadreccallid=${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 0 to
08 I
then originate a call via AMI to Local/0@outgoing-originate with
context set to outgoing-originate-rec and extension to 08.
The result will be something like this:
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-003f, cbrec-15605.wav,b) in new
stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-003f
-- Executing [h@outgoing-originate-rec:1]
AGI(SIP/upps-ccm-tq01-003e,
agi://l4574/ajpbxtest.agi?path=uploadreccallid=15605) in new stack
-- SIP/upps-ccm-tq01-003eAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid=15605 completed,
returning 0
-- Executing [h@outgoing-originate-rec-dev2:1]
AGI(SIP/upps-ccm-tq01-003f,
agi://4574/ajpbxtest.agi?path=uploadreccallid=) in new stack
-- SIP/upps-ccm-tq01-003fAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid= completed,
returning 0
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/upps-ccm-tq01-003f
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@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-0043, cbrec-15607.wav,b) in new
stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-0043
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0042, ) in new stack
== Spawn extension (outgoing-originate-rec-dev2, h, 1) exited
non-zero on
'SIP/upps-ccm-tq01-0042'
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0043, ) in new stack
== MixMonitor close filestream (mixed)
-- Executing [h@outgoing-originate-rec-dev2:2]
AGI(SIP/upps-ccm-tq01-0043,
agi://localhost/ajpbxtest.agi?path=uploadreccallid=) 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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Bharat Lalcheta
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
[asterisk-users] Recording with MixMonitor and AGI
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=uploadreccallid=${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 0 to 08 I then
originate a call via AMI to Local/0@outgoing-originate with context set
to outgoing-originate-rec and extension to 08.
The result will be something like this:
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-003f, cbrec-15605.wav,b) in new stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-003f
-- Executing [h@outgoing-originate-rec:1] AGI(SIP/upps-ccm-tq01-003e,
agi://l4574/ajpbxtest.agi?path=uploadreccallid=15605) in new stack
-- SIP/upps-ccm-tq01-003eAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid=15605 completed, returning 0
-- Executing [h@outgoing-originate-rec-dev2:1]
AGI(SIP/upps-ccm-tq01-003f,
agi://4574/ajpbxtest.agi?path=uploadreccallid=) in new stack
-- SIP/upps-ccm-tq01-003fAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid= completed, returning 0
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/upps-ccm-tq01-003f
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@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-0043, cbrec-15607.wav,b) in new stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-0043
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0042, ) in new stack
== Spawn extension (outgoing-originate-rec-dev2, h, 1) exited non-zero on
'SIP/upps-ccm-tq01-0042'
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0043, ) in new stack
== MixMonitor close filestream (mixed)
-- Executing [h@outgoing-originate-rec-dev2:2]
AGI(SIP/upps-ccm-tq01-0043,
agi://localhost/ajpbxtest.agi?path=uploadreccallid=) 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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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=uploadreccallid=${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 0
to 08 I then originate a call via AMI to
Local/0@outgoing-originate with context set
to outgoing-originate-rec and extension to 08.
The result will be something like this:
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-003f, cbrec-15605.wav,b) in new stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-003f
-- Executing [h@outgoing-originate-rec:1]
AGI(SIP/upps-ccm-tq01-003e,
agi://l4574/ajpbxtest.agi?path=uploadreccallid=15605) in new stack
-- SIP/upps-ccm-tq01-003eAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid=15605 completed,
returning 0
-- Executing [h@outgoing-originate-rec-dev2:1]
AGI(SIP/upps-ccm-tq01-003f,
agi://4574/ajpbxtest.agi?path=uploadreccallid=) in new stack
-- SIP/upps-ccm-tq01-003fAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid= completed,
returning 0
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/upps-ccm-tq01-003f
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@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-0043, cbrec-15607.wav,b) in new stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-0043
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0042, ) in new stack
== Spawn extension (outgoing-originate-rec-dev2, h, 1) exited
non-zero on 'SIP/upps-ccm-tq01-0042'
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0043, ) in new stack
== MixMonitor close filestream (mixed)
-- Executing [h@outgoing-originate-rec-dev2:2]
AGI(SIP/upps-ccm-tq01-0043,
agi://localhost/ajpbxtest.agi?path=uploadreccallid=) 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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
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 --
New to Asterisk? Join us for a live
Re: [asterisk-users] Recording with MixMonitor and AGI
As far as i understand your requirements, there is no need to use
macro for recording, You can directly call mixmonitor before Dial
application in your dialplan with required options. For transfer of
file, you are using AGI in h priority. However, you have to use
DeadAgi in h extension. As your channel already hangup, it can not
run on AGI.
Hope it will help you.
Regards,
Bharat Lalcheta
On Thu, Mar 7, 2013 at 8:51 PM, Henrik Westerberg
henrik.westerb...@ain.se wrote:
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=uploadreccallid=${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 0 to 08 I
then originate a call via AMI to Local/0@outgoing-originate with
context set to outgoing-originate-rec and extension to 08.
The result will be something like this:
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-003f, cbrec-15605.wav,b) in new stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-003f
-- Executing [h@outgoing-originate-rec:1]
AGI(SIP/upps-ccm-tq01-003e,
agi://l4574/ajpbxtest.agi?path=uploadreccallid=15605) in new stack
-- SIP/upps-ccm-tq01-003eAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid=15605 completed,
returning 0
-- Executing [h@outgoing-originate-rec-dev2:1]
AGI(SIP/upps-ccm-tq01-003f,
agi://4574/ajpbxtest.agi?path=uploadreccallid=) in new stack
-- SIP/upps-ccm-tq01-003fAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid= completed, returning 0
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/upps-ccm-tq01-003f
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@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-0043, cbrec-15607.wav,b) in new stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-0043
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0042, ) in new stack
== Spawn extension (outgoing-originate-rec-dev2, h, 1) exited non-zero on
'SIP/upps-ccm-tq01-0042'
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0043, ) in new stack
== MixMonitor close filestream (mixed)
-- Executing [h@outgoing-originate-rec-dev2:2]
AGI(SIP/upps-ccm-tq01-0043,
agi://localhost/ajpbxtest.agi?path=uploadreccallid=) 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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Bharat Lalcheta
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
