[asterisk-users] Problem with Caller ID when receiving hidden number in via DAHDI and redirecting out via SIP
Hi,
We have a system with both ISDN trunks and SIP. We receive incoming calls on
both but always dial out via SIP.
When dialing out the caller id is set like this:
exten = _X.,1,Set(CALLERID(num)=${CC_ORIGNUM})
exten = _X.,n,Set(CALLERID(name)=${CC_ORIGNAME})
exten = _X.,n,Dial(${CC_DIALSTRING}, 60, em)
This always works fine on SIP and on ISDN as well when the number is not hidden.
But for some reason the setting of the caller id does not work when receiving
calls from hidden numbers.
The from address in the outgoing SIP looks like this:
From: Anonymous sip:anonymous@anonymous.invalid
Does anyone know why this is happening, is there a way to go around it?
Regards,
Henrik
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Re: [asterisk-users] executing the h extension at the real hangup of the call
Ok, yes I find that strange as well. I will perform some tests on another
server.
/Henrik
Från: Gareth Blades
mailinglist+aster...@dns99.co.ukmailto:mailinglist+aster...@dns99.co.uk
Svara till: Asterisk Users Mailing List - Non-Commercial Discussion
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Datum: fredag 13 september 2013 13:53
Till: Asterisk Users Mailing List - Non-Commercial Discussion
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Ämne: Re: [asterisk-users] executing the h extension at the real hangup of the
call
On 13/09/13 12:31, Henrik Westerberg wrote:
Hi,
I am running Asterisk 11.3 with both SIP and ISDN. When dialing out (always
over SIP) I want to keep track of who answered and of the length of the call.
[outgoing-dev2]
exten = h,1,Agi(agi://localhost/ajpbxtest.agi?status=finished)
exten = _X.,1,NoOp(Will send call to ${CC_DIALSTRING})
exten = _X.,n,Dial(${CC_DIALSTRING}, 60, M(uploadpeer-dev2^${CC_CALLID})em)
exten =
_X.,n,Agi(agi://localhost/ajpbxtest.agi?status=faileddialstatus=${DIALSTATUS})
The h extension is called correctly when the call comes in over IP and when I
record the call. But when the call has come in over SIP the h extension is
called directly after the call is answered so all the call gets length 0 in my
own database.
I guess that I could record the calls and throw away the recordings afterwards.
In this way the RTP would stay on the server. But is there not a cleaner way to
get Asterisk to execute the h extension (or another possibility to fix a
callback somewhere) when the the Disconnect comes in over SIP?
I have no idea why you are seeing the h extension being run before the call
ends. Its not something I have ever seen happen.
Whether or not Asterisk stays in the RTP media path makes no difference as it
will always stay in the SIP signalling path and its that which controls the
call establishment and termination.
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[asterisk-users] executing the h extension at the real hangup of the call
Hi,
I am running Asterisk 11.3 with both SIP and ISDN. When dialing out (always
over SIP) I want to keep track of who answered and of the length of the call.
[outgoing-dev2]
exten = h,1,Agi(agi://localhost/ajpbxtest.agi?status=finished)
exten = _X.,1,NoOp(Will send call to ${CC_DIALSTRING})
exten = _X.,n,Dial(${CC_DIALSTRING}, 60, M(uploadpeer-dev2^${CC_CALLID})em)
exten =
_X.,n,Agi(agi://localhost/ajpbxtest.agi?status=faileddialstatus=${DIALSTATUS})
The h extension is called correctly when the call comes in over IP and when I
record the call. But when the call has come in over SIP the h extension is
called directly after the call is answered so all the call gets length 0 in my
own database.
I guess that I could record the calls and throw away the recordings afterwards.
In this way the RTP would stay on the server. But is there not a cleaner way to
get Asterisk to execute the h extension (or another possibility to fix a
callback somewhere) when the the Disconnect comes in over SIP?
Regards,
Henrik
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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,
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
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[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
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Re: [asterisk-users] Dialing out and recording
Yes I should really upgrade, just have to make sure that asterisk-java
will work properly with 1.8
/H
Den 2013-01-02 22:25 skrev Danny Nicholas da...@debsinc.com:
1.6.2 is a deader soldier than 1.4.X.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
Westerberg
Sent: Wednesday, January 02, 2013 3:20 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dialing out and recording
#2 works for me on Asterisk 1.8.12 when setting the header like this:
exten = _S,n,SipSetHeader(Diversion: ${CALLERID(rdnis)})
I haven't been able to make it work on 1.6 yet though, has anyone else?
/Henrik
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Wednesday, January 02, 2013 9:32 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Dialing out and recording
I have the same requirement, but it's important that the caller ID
information from the original caller is presented to the destination
and we announce the call before the transfer is complete. The carrier
requires a diversion header if the ANI is not one of our DIDs. Does
someone have experience with this working?
--
Two suggestions for you, Don. #1 if the Dial is Private the
announcement is taken care of. #2 I'm supposing that you could do a
SIP Header command before the Dial to resolve the diversion header
issue.
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[asterisk-users] Dialing out and recording
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten = _X.,1,Dial(SIP/${EXTEN},60,…)
exten = _X.,n,Agi(agi://localhost/aj.agi?action=……..)
I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and have AGI
control of it. Is there a way to do this or do I have to use G() and connect
the both ends to AGI separately and then bridging them before recording the
call?
Thanks for help.
Regards,
Henrik
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Re: [asterisk-users] Dialing out and recording
Thanks Danny I will try this.
/Henrik
Message: 12
Date: Wed, 2 Jan 2013 08:17:59 -0600
From: Danny Nicholas da...@debsinc.com
Subject: Re: [asterisk-users] Dialing out and recording
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: 001501cde8f3$f7d2b290$e77817b0$@debsinc.com
Content-Type: text/plain; charset=us-ascii
Put the AGI call in a macro context and add M(macro) to your Dial string.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
Westerberg
Sent: Wednesday, January 02, 2013 8:02 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dialing out and recording
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten = _X.,1,Dial(SIP/${EXTEN},60,.)
exten = _X.,n,Agi(agi://localhost/aj.agi?action=)
I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and
have
AGI control of it. Is there a way to do this or do I have to use G() and
connect the both ends to AGI separately and then bridging them before
recording the call?
Thanks for help.
Regards,
Henrik
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Re: [asterisk-users] Dialing out and recording
#2 works for me on Asterisk 1.8.12 when setting the header like this:
exten = _S,n,SipSetHeader(Diversion: ${CALLERID(rdnis)})
I haven't been able to make it work on 1.6 yet though, has anyone else?
/Henrik
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Wednesday, January 02, 2013 9:32 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Dialing out and recording
I have the same requirement, but it's important that the caller ID
information from the original caller is presented to the destination and
we
announce the call before the transfer is complete. The carrier requires
a
diversion header if the ANI is not one of our DIDs. Does someone have
experience with this working?
--
Two suggestions for you, Don. #1 if the Dial is Private the
announcement is taken care of. #2 I'm supposing that you could do a SIP
Header command before the Dial to resolve the diversion header issue.
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Re: [asterisk-users] async agi question
Thanks, I was not familiar with this application. /Henrik Kevin P. Fleming skrev: Henrik Westerberg wrote: Yes, this works good for me. A StopIO feature would of course be cleaner but this certainly does the trick. The ExternalIVR interface, while not quite as feature-filled as AGI, does in fact work in a true non-blocking fashion, and supports exactly what you are looking for. In fact, needing to be able to stop playback of prompts asynchronously was the primary reason it was developed. ___ -- 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
[asterisk-users] async agi question
Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven't been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which I have missed? Or could someone give me hints on how I could implement this in the res_agi.c The command asyncagi break does stop ongoing playing but also breaks the async agi control. I only want the first. Thanks in advance, /Henrik ___ -- 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
Re: [asterisk-users] async agi question
Hi Moy, Thank you for your quick answer. Also thanks for implementing the great async agi functionality! Yes, this works good for me. A StopIO feature would of course be cleaner but this certainly does the trick. Regards, Henrik Moises Silva skrev: Hello Henrik, I have not used Asterisk from a user perspective lately, but, when I added the async agi functionality, I used to control this using a "manager redirect" action to the same priority where the channel calls async agi, that will work like a break that re-enters the async agi loop . This, of course, requires you to save the state of the channel somehow in your program to "remember" that the next time that channel calls async agi the sound was already played and such. http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect Let me know if that does not work for you and we can probably write something in res_agi.c Moy On Fri, Dec 5, 2008 at 3:01 AM, Henrik Westerberg [EMAIL PROTECTED] wrote: Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven't been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which I have missed? Or could someone give me hints on how I could implement this in the res_agi.c The command "asyncagi break" does stop ongoing playing but also breaks the async agi control. I only want the first. Thanks in advance, /Henrik ___ -- 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 -- Med vnliga hlsningar / Best Regards Henrik Westerberg Software Developer Aurora Innovation AB Vallongatan 1, 752 28 Uppsala, Sweden direct: +46 18 19 44 58 mobile: +46 703 28 98 40 email: [EMAIL PROTECTED] www.aurorainnovation.se ___ -- 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
[asterisk-users] pri rdnis found as Facility but not set
Hi, I'm running asterisk with a PRI. But I can't get hold of the rdnis number. When running pri debug I can see the true rdnis number as Facility, the number 703289840 as shown below. Is it possible to get hold of this value in some way from extensions.conf? Or is it necessary to modify the source for asterisk, in that case does someone know where and how? Thanks in advance, Henrik Protocol Discriminator: Q.931 (8) len=79 Call Ref: len= 2 (reference 40/0x28) (Originator) Message type: SETUP (5) [a1] Sending Complete (len= 1) [04 03 90 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a1 83 8a] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 10 ] [1c 26 91 a1 23 02 02 00 80 02 01 0f 30 1a 02 01 01 0a 01 02 a1 12 a0 10 a1 0e 0a 01 02 12 09 37 30 33 32 38 39 38 34 30] Facility (len=40, codeset=0) [ 0x91, 0xa1, 0x23, 0x02, 0x02, 0x00, 0x80, 0x02, 0x01, 0x0f, '0', 0x1a, 0x02, 0x01, 0x01, 0x0a, 0x01, 0x02, 0xa1, 0x12, 0xa0, 0x10, 0xa1, 0x0e, 0x0a, 0x01, 0x02, 0x12, 0x09, '703289840' ] [1e 02 84 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 0a 21 83 31 38 31 33 34 32 35 35] Calling Number (len=12) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '18134255' ] [70 05 c1 38 35 35 36] Called Number (len= 7) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '8556' ] -- Making new call for cr 40 -- Processing Q.931 Call Setup -- Processing IE 161 (cs0, Sending Complete) -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 28 (cs0, Facility) Handle Q.932 ROSE Invoke component -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) Aug 17 16:36:39 WARNING[31243]: chan_zap.c:8379 pri_dchannel: PRI_EVENT_RING Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 40/0x28) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 8a] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 10 ] -- Accepting call from '18134255' to '8556' on channel 0/10, span 1 -- Executing Answer(Zap/10-1, ) in new stack Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 40/0x28) (Terminator) Message type: CONNECT (7) [18 03 a9 83 8a] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 10 ] [1e 02 81 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Executing NoOp(Zap/10-1, name: ) in new stack -- Executing NoOp(Zap/10-1, number: 18134255) in new stack -- Executing NoOp(Zap/10-1, ani: 18134255) in new stack -- Executing NoOp(Zap/10-1, dnid: 8556) in new stack -- Executing NoOp(Zap/10-1, rdnis: ) in new stack -- Executing Goto(Zap/10-1, test|1) in new stack -- Goto (default,test,1) -- Executing Answer(Zap/10-1, ) in new stack -- Executing Wait(Zap/10-1, 1) in new stack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
