Re: [asterisk-users] ControlPlayback's options

[Johan Wilfer]

On 2011-06-04 13:38, virendra bhati wrote:

Hi Johan Wilfer,

Thanks for your reply. On the basis of your provided code I made all 
things into extensions.conf. But i have an small issue on which I need 
your attention again.

in below context what's  ${tz} ? Is this time zone value or else?
Yes, I store the calls timezone in tis variable before the code sample 
you got.


and another things is what is the use of SayUnixTime(${time},${tz},d 
'digits/of' B);

this function in such case?


I've implemented 5 as a pause-button on the phone. This context handles 
this by playing a

prompt that you have pause the recording and  and .

This is repeated untill the user presses a key on the keypad.


context conference_play_recordings_
conference_paused {
announce => {
  Set(time=$[${epoch_start}+${position}/1000]);
  while(true) {
WaitExten(1);
Background(conf_playrec_pause_part1);
SayUnixTime(${time},${tz},kM);
Background(conf_playrec_pause_part2);
SayUnixTime(${time},${tz},d 'digits/of' B);
Background(conf_playrec_pause_part3);
WaitExten(5);
  }
}

one thing which is also confusing is that what is the meaning or use 
of such lines in this application.


ControlPlayback(${filename},6,3,1,*#2456790,,,o(${position}))

${filename} is the file you want to play.
6 is 60 seconds to skip.
3 is to use 3 as forward 60 seconds
1 to to use 1 as rewind  60 seconds
*#2456790 is used as stop buttons (and handled by the dialplan)
o() is a option to go to a specific position in the file
${position} is the variable that hold the current position of the playback.

To get more details use the following command:
asterisk*CLI> core show application ControlPlayback

Displays:
  -= Info about application 'ControlPlayback' =-

[Synopsis]
Play a file with fast forward and rewind.

[Description]
This application will play back the given .
It sets the following channel variables upon completion:
${CPLAYBACKSTATUS}: Contains the status of the attempt as a text string
SUCCESS
USERSTOPPED
ERROR
${CPLAYBACKOFFSET}: Contains the offset in ms into the file where playback
was at when it stopped. '-1' is end of file.
${CPLAYBACKSTOPKEY}: If the playback is stopped by the user this variable
contains the key that was pressed.

[Syntax]
ControlPlayback(filename[,skipms[,ff[,rew[,stop[,pause[,restart[,options]]])

[Arguments]
skipms
This is number of milliseconds to skip when rewinding or fast-fo
rwarding.
ff
Fast-forward when this DTMF digit is received. (defaults to '#')
rew
Rewind when this DTMF digit is received. (defaults to '*')
stop
Stop playback when this DTMF digit is received.
pause
Pause playback when this DTMF digit is received.
restart
Restart playback when this DTMF digit is received.
options
o(time):
time - Start at  ms from the beginning of the
file.

[See Also]
Not available


/Johan


Please put some light on these too.

On Wed, Jun 1, 2011 at 1:50 AM, Johan Wilfer > wrote:


On 2011-05-30 14:32, virendra bhati wrote:

Hi List,

Asterisk 's *ControlPlayback* will used for play any recorded
file as an audio player. Is it possible that we can use it for
multiple forward and rewind ?

ex:-
original: ControlPlayback(filename,skipms,ff,rew,stop,pause)
expected

ControlPlayback(filename,skip1,skip2,skip3,forward1,rewind1,forward2,rewind2,forward3,rewind3,stop,pause)
:


Yes, you can use the CPLAYBACKSTATUS, CPLAYBACKOFFSET and
CPLAYBACKSTOPKEY variables to get this behavior.
All you have to do is to list the additional keys and stop keys
and implement this in your dialplan...

I've attached some ael I use for this to implement 1 and 3 as 1
minute rewind/forward. 4 and 6 as 5 minutes rewind/forward and 7
and 9 as 15 minutes.
5 I use as the pause key, and */# to switch recording.

Greetings,
Johan Wilfer




  context conference_play_recordings_conference_connect {
playrec_intro => {
  Set(position=0);
  goto play,1;
}

play => {
  while (true) {
if (${position}==-1) { goto recording_end,1; }

//rewind 5 seconds after every action (so the user doesn't
feel lost...)
Set(position=$[${position}-5000]);
if (${position} < 0) { Set(position=0); }

   
ControlPlayback(${filename},6,3,1,*#2456790,,,o(${position}));

Set(position=${CPLAYBACKOFFSET});

if (${CPLAYBACKSTATUS}==ERROR) {
  Playback(pbx_error_500);
  Playback(pbx_endcall);
  Wait(2);
  Hangup();
}

//If stopped by user
if (${CPLAYBACKSTATUS}==USERSTOPPED) {
  if (!${ISNULL(${CPLAYBACKSTOPKEY})}) { goto
${CPLAYBACKSTOPKEY},1; }
}
  }
}

recording_end => {
  

[asterisk-users] Blind transfer issue on Asterisk 1.8.4.2

[Bart Coninckx]

Hi all,

when doing a blind transfer using the keys defined in features.conf, we 
hear a confirmation of the attempt to blindly transfer, followed by an 
invalid extension message.


The console says this:

[Jun  4 22:30:31] VERBOSE[11301] res_musiconhold.c: -- Started music 
on hold, class 'default', on SIP/570-0006
[Jun  4 22:30:31] VERBOSE[11301] file.c: --  
Playing 'pbx-transfer.gsm' (language 'nl')
[Jun  4 22:30:32] WARNING[11301] features.c: Extension '53' does not 
exist in context 'transfer_context,570,1'
[Jun  4 22:30:32] VERBOSE[11301] file.c: --  
Playing 'pbx-invalid.gsm' (language 'nl')
[Jun  4 22:30:34] VERBOSE[11301] res_musiconhold.c: -- Stopped music 
on hold on SIP/570-0006


Mind you, the entered extension is "531", so it seems part of the entry 
is cut off. Sometimes it shows just "5". It is as if featuredigittimeout 
(set to 2000) is not taken into account.



Thx!!


B.




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[asterisk-users] asterisk 1.6 - 511 Command not permitted causing high CPU usage

[vip killa]

http://pastebin.com/vxGM2n5j

We are getting those errors 100x per second in console when AGI set debug is
on
It is causing extremely high CPU usage, we've tried asterisk version
1.6.1.22 and 1.6.2.18
It seems the problem is worse in 1.6.2.18
Can someone advise how to fix this? Thank you.
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Re: [asterisk-users] benefits of asterisk 1.8

[dotnetdub]

On 3 June 2011 22:41, Hans Witvliet  wrote:

> On Fri, 2011-06-03 at 09:07 +0100, Ishfaq Malik wrote:
> > Are you suggesting that there are no bugs in 1.4 or 1.6?
>
> I presume that you are aware of the fact that it is impossible to prove
> the absence of "bugs" in any piece of software
> You might not have detected them yet.
> Furthermore behaviour that might have been coded on purpose, can be
> considered "eroneously" some time later.
>
> > Currently there seems to be a fear of 1.8. We're about to put it into
> > production and yes, we've had issues with it, mostly due to the fact we
> > use RealTime, but before you change anything it is always advisable to
> > test the hell out of it.
> >
> > To anyone who is thinking of moving to 1.8 the question is not, 'is it
> > stable?'. The question is, 'have I comprehensively tested it to show
> > that it is suitable for my needs?'
>
> If you put it into production, test at least the functions that you are
> going to use. There might (and probably will) problems in the code, but
> as long as it does not bother you, so what?
>
>

See this thread here about Asterisk 1.8 - and Digium's view on the matter.

http://nerdvittles.com/?p=743
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Re: [asterisk-users] benefits of asterisk 1.8

[vip killa]

digium, eat your dog food!

On Sun, Jun 5, 2011 at 11:08 AM, dotnetdub  wrote:

>
>
> On 3 June 2011 22:41, Hans Witvliet  wrote:
>
>> On Fri, 2011-06-03 at 09:07 +0100, Ishfaq Malik wrote:
>> > Are you suggesting that there are no bugs in 1.4 or 1.6?
>>
>> I presume that you are aware of the fact that it is impossible to prove
>> the absence of "bugs" in any piece of software
>> You might not have detected them yet.
>> Furthermore behaviour that might have been coded on purpose, can be
>> considered "eroneously" some time later.
>>
>> > Currently there seems to be a fear of 1.8. We're about to put it into
>> > production and yes, we've had issues with it, mostly due to the fact we
>> > use RealTime, but before you change anything it is always advisable to
>> > test the hell out of it.
>> >
>> > To anyone who is thinking of moving to 1.8 the question is not, 'is it
>> > stable?'. The question is, 'have I comprehensively tested it to show
>> > that it is suitable for my needs?'
>>
>> If you put it into production, test at least the functions that you are
>> going to use. There might (and probably will) problems in the code, but
>> as long as it does not bother you, so what?
>>
>>
>
> See this thread here about Asterisk 1.8 - and Digium's view on the matter.
>
> http://nerdvittles.com/?p=743
>
>
>
>
>
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Re: [asterisk-users] asterisk 1.6 - 511 Command not permitted causing high CPU usage

[Steve Edwards]

On Sun, 5 Jun 2011, vip killa wrote:


http://pastebin.com/vxGM2n5j

We are getting those errors 100x per second in console when AGI set 
debug is on



Can someone advise how to fix this? Thank you.


Don't request 'WAIT FOR DIGIT 1000' from a dead channel.

Don't ignore the error from 'WAIT FOR DIGIT 1000'

Don't loop on the error.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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[asterisk-users] broken SVN asterisk 1.8 ?

[satish patel]

Hey guys!

I have just download latest SVN Revision 322051 and compile and install but my 
asterisk -V showing still old version :( is it broken ?

/usr/sbin/asterisk -V
Asterisk SVN-branch-1.8-r321926

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[asterisk-users] DTMF issue in app_konference using with asterisk 1.8.3.2

[Krishna Sumanth Chava]

Hi,

I have a requirement where the DTMF entered by a member in konference is
passed on to the other members.

But the DTMF is not being recognized, when checked the events from manager
API, I do see DTMF event being passed, but some how it is not passed to
other members.

This tells me - may be it is not an asterisk issue, but more a konference
application issue.

Is this not supported by app_konference or am i missing any thing specific
in my configuration? I am using the flags (Mx) for the konference
application.

I am using asterisk 1.8.3.2 and app_konference 1.7.
I also tried changing DTMF to "1" in the Makefile of konference application.


Wondering if any other member in the group is having a similar issue with
app_konference and asterisk 1.8.3.2 and any suggestions would be
appreciated?

Thanks,
Krishna
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Re: [asterisk-users] ControlPlayback's options

[virendra bhati]

Hi John Wilfer,

Thanks for replay. Now all things is working on asterisk 1.6.2.18 version.
But When I try the same thing on Asterisk 1.4.X then facing problem.

Is this the problem of  ControlPlayback 's option fields of asterisk 1.4.X
in this version have option P(jumping) not O(time) ?
Is there any way by which we will implement like by upload ControlPlayback
from asterisk 1.6 to 1.4 or else ?
ControlPlayback(filename[,skipms[,ff[,rew[,stop[,pause[,restart[,options]]])

On Sun, Jun 5, 2011 at 2:16 PM, Johan Wilfer  wrote:

>  On 2011-06-04 13:38, virendra bhati wrote:
>
> Hi Johan Wilfer,
>
> Thanks for your reply. On the basis of your provided code I made all things
> into extensions.conf. But i have an small issue on which I need your
> attention again.
> in below context what's  ${tz} ? Is this time zone value or else?
>
> Yes, I store the calls timezone in tis variable before the code sample you
> got.
>
>
>  and another things is what is the use of  SayUnixTime(${time},${tz},d
> 'digits/of' B);
> this function in such case?
>
>
> I've implemented 5 as a pause-button on the phone. This context handles
> this by playing a
> prompt that you have pause the recording and  and .
>
> This is repeated untill the user presses a key on the keypad.
>
>
> context conference_play_recordings_
> conference_paused {
> announce => {
>   Set(time=$[${epoch_start}+${position}/1000]);
>   while(true) {
> WaitExten(1);
> Background(conf_playrec_pause_part1);
> SayUnixTime(${time},${tz},kM);
> Background(conf_playrec_pause_part2);
> SayUnixTime(${time},${tz},d 'digits/of' B);
> Background(conf_playrec_pause_part3);
> WaitExten(5);
>   }
> }
>
> one thing which is also confusing is that what is the meaning or use of
> such lines in this application.
>
> ControlPlayback(${filename},6,3,1,*#2456790,,,o(${position}))
>
> ${filename} is the file you want to play.
> 6 is 60 seconds to skip.
> 3 is to use 3 as forward 60 seconds
> 1 to to use 1 as rewind  60 seconds
> *#2456790 is used as stop buttons (and handled by the dialplan)
> o() is a option to go to a specific position in the file
> ${position} is the variable that hold the current position of the playback.
>
> To get more details use the following command:
> asterisk*CLI> core show application ControlPlayback
>
> Displays:
>   -= Info about application 'ControlPlayback' =-
>
> [Synopsis]
> Play a file with fast forward and rewind.
>
> [Description]
> This application will play back the given .
> It sets the following channel variables upon completion:
> ${CPLAYBACKSTATUS}: Contains the status of the attempt as a text string
> SUCCESS
> USERSTOPPED
> ERROR
> ${CPLAYBACKOFFSET}: Contains the offset in ms into the file where playback
> was at when it stopped. '-1' is end of file.
> ${CPLAYBACKSTOPKEY}: If the playback is stopped by the user this variable
> contains the key that was pressed.
>
> [Syntax]
>
> ControlPlayback(filename[,skipms[,ff[,rew[,stop[,pause[,restart[,options]]])
>
> [Arguments]
> skipms
> This is number of milliseconds to skip when rewinding or fast-fo
> rwarding.
> ff
> Fast-forward when this DTMF digit is received. (defaults to '#')
> rew
> Rewind when this DTMF digit is received. (defaults to '*')
> stop
> Stop playback when this DTMF digit is received.
> pause
> Pause playback when this DTMF digit is received.
> restart
> Restart playback when this DTMF digit is received.
> options
> o(time):
> time - Start at  ms from the beginning of the
> file.
>
> [See Also]
> Not available
>
>
> /Johan
>
>
> Please put some light on these too.
>
> On Wed, Jun 1, 2011 at 1:50 AM, Johan Wilfer  wrote:
>
>>  On 2011-05-30 14:32, virendra bhati wrote:
>>
>> Hi List,
>>
>> Asterisk 's *ControlPlayback* will used for play any recorded file as an
>> audio player. Is it possible that we can use it for multiple forward and
>> rewind ?
>>
>> ex:-
>> original: ControlPlayback(filename,skipms,ff,rew,stop,pause)
>> expected
>> ControlPlayback(filename,skip1,skip2,skip3,forward1,rewind1,forward2,rewind2,forward3,rewind3,stop,pause)
>> :
>>
>>
>>  Yes, you can use the CPLAYBACKSTATUS, CPLAYBACKOFFSET and
>> CPLAYBACKSTOPKEY variables to get this behavior.
>> All you have to do is to list the additional keys and stop keys and
>> implement this in your dialplan...
>>
>> I've attached some ael I use for this to implement 1 and 3 as 1 minute
>> rewind/forward. 4 and 6 as 5 minutes rewind/forward and 7 and 9 as 15
>> minutes.
>> 5 I use as the pause key, and */# to switch recording.
>>
>> Greetings,
>> Johan Wilfer
>>
>> 
>>
>>
>>   context conference_play_recordings_conference_connect {
>> playrec_intro => {
>>   Set(position=0);
>>   goto play,1;
>> }
>>
>> play => {
>>   while (true) {
>> if (${position}==-1) { goto recording_end,1; }
>>
>> //rewind 5 seconds after every action (so 

[asterisk-users] Asterisk users Calculation

[Khaled W. Chehab]

Dears

 

I already read most of post on asterisk group and
(http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning)

But I could not find a calculator 

1-Is there a calculator I can download for that 

2-What I the maximum simultaneous  calls that can asterisk handle using CPU
3.0 MHZ and 4GB ram

With rtp g729 and  there is no codec transcoding ,

3-And what is the number of simultaneous calls if I use direct  RTP
(Canreinvite=no /Directrt=yes)

 

 

Regards

 

 

Khaled  Chehab

   NGN Eng.

 

Description: xplorium 

 Operations Office - Lebanon

 Office : +961 1 868686 ext 115

 Mobile: +961 3 045212

 E-mail:   kche...@xplorium.com

 MSN ID :khalidche...@hotmail.com  

 Web Site: http://www.xplorium.com

 

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Re: [asterisk-users] Asterisk users Calculation

[Steve Edwards]

On Sun, 5 Jun 2011, Khaled W. Chehab wrote:


1-Is there a calculator I can download for that

2-What I the maximum simultaneous calls that can asterisk handle using 
CPU 3.0 MHZ and 4GB ram With rtp g729 and there is no codec transcoding 

3-And what is the number of simultaneous calls if I use direct RTP 
(Canreinvite=no /Directrt=yes)


1) No. Because every case is a bit different and nobody has taken the time 
to research and document it.


2) In the 'hundreds.' I have a 5 yr old 3.4 Xeon server with 2GB of ram 
running all kinds of AGIs that handles 300 simultaneous ULAW calls without 
issue and without any 'tuning.' The Asterisk process uses less than 100MB

so more GBs means nothing.

3) Probably in the thousands depending on what those calls are doing. 
(Just guessing here because I have no experience with this configuration.)


Would a SIP server like OpenSIPS be a better platform choice?

More details will yield better responses.

--
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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] broken SVN asterisk 1.8 ?

[Barry Miller]

On Sun, Jun 05, 2011 at 04:18:25PM +, satish patel wrote:
> 
> Hey guys!
> 
> I have just download latest SVN Revision 322051 and compile and install but 
> my asterisk -V showing still old version :( is it broken ?
> 
> /usr/sbin/asterisk -V
> Asterisk SVN-branch-1.8-r321926

asterisk -V shows the last changed revision in the build.

To see the difference, try:

   cd asterisk-src-dir
   svnversion
   svnversion -c

-- 
Barry 

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Re: [asterisk-users] Asterisk users Calculation

[Sherwood McGowan]

May I add...I still have documented cases of asterisk 1.4.x running ulaw with 
no transcoding and running 2k+ concurrent calls on a CentOS 4(5?, fuzzy) 
machine with 2ghz CPU and 2gb ram

Sent from my iPhone

On Jun 5, 2011, at 3:02 PM, Steve Edwards  wrote:

> On Sun, 5 Jun 2011, Khaled W. Chehab wrote:
> 
>> 1-Is there a calculator I can download for that
>> 2-What I the maximum simultaneous calls that can asterisk handle using CPU 
>> 3.0 MHZ and 4GB ram With rtp g729 and there is no codec transcoding 3-And 
>> what is the number of simultaneous calls if I use direct RTP (Canreinvite=no 
>> /Directrt=yes)
> 
> 1) No. Because every case is a bit different and nobody has taken the time to 
> research and document it.
> 
> 2) In the 'hundreds.' I have a 5 yr old 3.4 Xeon server with 2GB of ram 
> running all kinds of AGIs that handles 300 simultaneous ULAW calls without 
> issue and without any 'tuning.' The Asterisk process uses less than 100MB
> so more GBs means nothing.
> 
> 3) Probably in the thousands depending on what those calls are doing. (Just 
> guessing here because I have no experience with this configuration.)
> 
> Would a SIP server like OpenSIPS be a better platform choice?
> 
> More details will yield better responses.
> 
> -- 
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
> 
> --
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>  http://www.asterisk.org/hello
> 
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> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] broken SVN asterisk 1.8 ?

[Satish Patel]

Thanks but they should change svn revesion number change in file.

--
Sent from my iPhone

On Jun 5, 2011, at 7:13 PM, Barry Miller   
wrote:



On Sun, Jun 05, 2011 at 04:18:25PM +, satish patel wrote:


Hey guys!

I have just download latest SVN Revision 322051 and compile and  
install but my asterisk -V showing still old version :( is it  
broken ?


/usr/sbin/asterisk -V
Asterisk SVN-branch-1.8-r321926


asterisk -V shows the last changed revision in the build.

To see the difference, try:

  cd asterisk-src-dir
  svnversion
  svnversion -c

--
Barry

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[asterisk-users] AGI STREAM FILE not working?

[A E [Gmail]]

Hello,
using 1.8.4. using a very simple local AGI script in bash which has only one
line in it:

echo -e 'STREAM FILE welcome 123 \n'

dialplan:
exten => 5150,1,Answer()
  same => n,Set(CHANNEL(language)=en_AU)
  same => n,AGI(testagi.sh)
  same => n,Hangup

console output:
-- Executing [5150@AllPhones:1] Answer("SIP/PBX-0024", "") in new
stack
-- Executing [5150@AllPhones:2] Set("SIP/PBX-0024",
"CHANNEL(language)=en_AU") in new stack
-- Executing [5150@AllPhones:3] AGI("SIP/PBX-0024", "testagi.sh") in
new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/testagi.sh
-- Playing 'welcome' (escape_digits=1) (sample_offset 0)
-- AGI Script testagi.sh completed, returning 0
-- Executing [5150@AllPhones:4] Hangup("SIP/PBX-0024", "") in new
stack
  == Spawn extension (AllPhones, 5150, 4) exited non-zero on
'SIP/PBX-0024'

But nothing happens...as in even when it says that it's playing the file (as
verified in the asterisk 'full' log), I hear nothing on the phone

What gives? spent 2 hrs Googling but nothing! :(

Thx
\A
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Re: [asterisk-users] Asterisk users Calculation

[Satish Barot]

It would be a great help to others(including me) if those using 1.8.X can
provide some details on hardware configurations,features they have
implemented on it and some sort of load testing results.

Thanks,
[SATISH]

On Mon, Jun 6, 2011 at 6:28 AM, Sherwood McGowan  wrote:

> May I add...I still have documented cases of asterisk 1.4.x running ulaw
> with no transcoding and running 2k+ concurrent calls on a CentOS 4(5?,
> fuzzy) machine with 2ghz CPU and 2gb ram
>
>
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[asterisk-users] issues.asterisk.org

[Jeremy Kister]

i'm trying to review issues that i'm monitoring and/or have reported at 
http://issues.asterisk.org


when I click on 'My View' or 'View Issues' I get an error:
APPLICATION ERROR #401

Database query failed. Error received from database was #1142: DELETE 
command denied to user 'mantisreadonly'@'localhost' for table 
'mantis_tokens_table' for the query: DELETE FROM mantis_tokens_table 
WHERE '2011-06-06 00:03:56' > expiry.



Are tickets that I had set up for monitoring on mantis going to be 
automatically monitored in jira ?


similarly, are tickets that I reported in mantis going to show as me 
being the reporter in jira?  or are the tickets going to stay in mantis 
until they are resolved and never make it into jira ?



--

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http://jeremy.kister.net./

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Re: [asterisk-users] issues.asterisk.org

[Jeremy Kister]

On 6/6/2011 1:08 AM, Jeremy Kister wrote:

similarly, are tickets that I reported in mantis going to show as me
being the reporter in jira?  or are the tickets going to stay in mantis
until they are resolved and never make it into jira ?


after some more clicking, i see the answer to this one; nevermind.


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Re: [asterisk-users] AGI STREAM FILE not working?

[mahesh katta]

On Mon, Jun 6, 2011 at 9:42 AM, A E [Gmail]  wrote:

> Hello,
> using 1.8.4. using a very simple local AGI script in bash which has only
> one line in it:
>
> echo -e 'STREAM FILE welcome 123 \n'
>
> dialplan:
> exten => 5150,1,Answer()
>   same => n,Set(CHANNEL(language)=en_AU)
>   same => n,AGI(testagi.sh)
>
same => n,Hangup
>
Try this below dilaplan

> exten => 5150,1,Answer()
>   same => n,Set(CHANNEL(language)=en_AU)
>   same => n,AGI(testagi)
> same => n,Hangup
>


> console output:
> -- Executing [5150@AllPhones:1] Answer("SIP/PBX-0024", "") in new
> stack
> -- Executing [5150@AllPhones:2] Set("SIP/PBX-0024",
> "CHANNEL(language)=en_AU") in new stack
> -- Executing [5150@AllPhones:3] AGI("SIP/PBX-0024", "testagi.sh")
> in new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/testagi.sh
> -- Playing 'welcome' (escape_digits=1) (sample_offset 0)
> -- AGI Script testagi.sh completed, returning 0
> -- Executing [5150@AllPhones:4] Hangup("SIP/PBX-0024", "") in new
> stack
>   == Spawn extension (AllPhones, 5150, 4) exited non-zero on
> 'SIP/PBX-0024'
>
> But nothing happens...as in even when it says that it's playing the file
> (as verified in the asterisk 'full' log), I hear nothing on the phone
>
> What gives? spent 2 hrs Googling but nothing! :(
>
> Thx
> \A
>
>
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> _
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
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Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
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Web http://www.buzzworks.com
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Re: [asterisk-users] AGI STREAM FILE not working?

[Steve Edwards]

On Mon, 6 Jun 2011, A E [Gmail] wrote:

Hello,using 1.8.4. using a very simple local AGI script in bash which 
has only one line in it:


echo -e 'STREAM FILE welcome 123 \n'

What gives? spent 2 hrs Googling but nothing! :(


Maybe 1.5 hrs should have been spent reading :)

One line does not an AGI make.

AGI is an interface. It consists of reading the AGI environment from STDIN 
and then, writing requests on STDOUT and reading the response from STDIN.


If you don't follow these 3 steps in order (steps 2 and 3 can be repeated) 
then your program has violated the protocol and will not function reliably 
if at all.


Please use an existing AGI library for the language of your choice. Nobody 
gets it right the first time.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] AGI STREAM FILE not working?

[A E [Gmail]]

On Mon, Jun 6, 2011 at 12:12 AM, A E [Gmail]  wrote:

> Hello,
> using 1.8.4. using a very simple local AGI script in bash which has only
> one line in it:
>
> echo -e 'STREAM FILE welcome 123 \n'
>
> dialplan:
> exten => 5150,1,Answer()
>   same => n,Set(CHANNEL(language)=en_AU)
>   same => n,AGI(testagi.sh)
>   same => n,Hangup
>
> console output:
> -- Executing [5150@AllPhones:1] Answer("SIP/PBX-0024", "") in new
> stack
> -- Executing [5150@AllPhones:2] Set("SIP/PBX-0024",
> "CHANNEL(language)=en_AU") in new stack
> -- Executing [5150@AllPhones:3] AGI("SIP/PBX-0024", "testagi.sh")
> in new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/testagi.sh
> -- Playing 'welcome' (escape_digits=1) (sample_offset 0)
> -- AGI Script testagi.sh completed, returning 0
> -- Executing [5150@AllPhones:4] Hangup("SIP/PBX-0024", "") in new
> stack
>   == Spawn extension (AllPhones, 5150, 4) exited non-zero on
> 'SIP/PBX-0024'
>
> But nothing happens...as in even when it says that it's playing the file
> (as verified in the asterisk 'full' log), I hear nothing on the phone
>
> What gives? spent 2 hrs Googling but nothing! :(
>
> Thx
> \A
>
>  Bizarre, I found a bunch of other agi scripts in the default
directory...modified the agi-test.agi (perl script) so it played my file, no
joy! then I used a php script I found somewhere else asa tutorial to writing
AGI scripts in php, modified that to play my script and it works. I don't
get it. esp. when everything (with agi debug set on) looks exactly the same
with my bash script and this php script except that with the php script, I
see this ONE line that's extra


AGI Rx << STREAM FILE welcome #
-- Playing 'welcome' (escape_digits=#) (sample_offset 0)
AGI Tx >> 200 result=35 endpos=87200

that I don't see with my bash script which does this


AGI Rx << STREAM FILE welcome #

-- Playing 'welcome' (escape_digits=#) (sample_offset 0)

-- AGI Script streamcontact.sh completed, returning 0

 -- Executing [5150@AllPhones:5] Hangup("SIP/PBX-002e", "") in new
stack

So confused!!
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Re: [asterisk-users] AGI STREAM FILE not working?

[A E [Gmail]]

On Mon, Jun 6, 2011 at 2:06 AM, mahesh katta wrote:

>
>
> On Mon, Jun 6, 2011 at 9:42 AM, A E [Gmail]  wrote:
>
>> Hello,
>> using 1.8.4. using a very simple local AGI script in bash which has only
>> one line in it:
>>
>> echo -e 'STREAM FILE welcome 123 \n'
>>
>> dialplan:
>> exten => 5150,1,Answer()
>>   same => n,Set(CHANNEL(language)=en_AU)
>>   same => n,AGI(testagi.sh)
>>
> same => n,Hangup
>>
> Try this below dilaplan
>
>> exten => 5150,1,Answer()
>>   same => n,Set(CHANNEL(language)=en_AU)
>>   same => n,AGI(testagi)
>> same => n,Hangup
>>
>
No deal. Doesn't find the AGI script

>
>
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Re: [asterisk-users] AGI STREAM FILE not working?

[A E [Gmail]]

On Mon, Jun 6, 2011 at 2:06 AM, Steve Edwards wrote:

> On Mon, 6 Jun 2011, A E [Gmail] wrote:
>
>  Hello,using 1.8.4. using a very simple local AGI script in bash which has
>> only one line in it:
>>
>> echo -e 'STREAM FILE welcome 123 \n'
>>
>> What gives? spent 2 hrs Googling but nothing! :(
>>
>
> Maybe 1.5 hrs should have been spent reading :)
>

touche ;)

>
> One line does not an AGI make.
>
> Did you just pull a 'Yoda' Steve?


> AGI is an interface. It consists of reading the AGI environment from STDIN
> and then, writing requests on STDOUT and reading the response from STDIN.
>
> Right! I did read that, the problem is how do I do this in bash?? I tried
read the result in and just post a Noop kind of a thing just to tell that I
read something, but it didn't help. I also explicitly did that in the perl
script, but doesn't work. It only works in PHP.


> If you don't follow these 3 steps in order (steps 2 and 3 can be repeated)
> then your program has violated the protocol and will not function reliably
> if at all.
>
> Please use an existing AGI library for the language of your choice. Nobody
> gets it right the first time.
>
>
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Re: [asterisk-users] AGI STREAM FILE not working?

[Steve Edwards]

On Mon, Jun 6, 2011 at 2:06 AM, Steve Edwards 
 wrote:


AGI is an interface. It consists of reading the AGI environment from 
STDIN and then, writing requests on STDOUT and reading the response from 
STDIN.


On Mon, 6 Jun 2011, A E [Gmail] wrote:

Right! I did read that, the problem is how do I do this in bash?? I 
tried read the result in and just post a Noop kind of a thing just to 
tell that I read something, but it didn't help. I also explicitly did 
that in the perl script, but doesn't work. It only works in PHP.


Bash would probably be my last choice of language to write an AGI with.

Personally, I use C because it is my sharpest tool and because you can 
execute hundreds of AGIs written in C in the time it takes to load the 
Perl or PHP interpreter and parse your script.


I suspect that the problems you are experiencing with Perl may have 
something to do with flushing STDOUT or reading the complete response from 
STDIN.


I strongly suggest using an existing library for the language of your 
choice.


--
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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] AGI STREAM FILE not working?

[A E [Gmail]]

On Mon, Jun 6, 2011 at 2:26 AM, Steve Edwards wrote:

>  On Mon, Jun 6, 2011 at 2:06 AM, Steve Edwards 
>> wrote:
>>
>
>  AGI is an interface. It consists of reading the AGI environment from STDIN
>> and then, writing requests on STDOUT and reading the response from STDIN.
>>
>
> On Mon, 6 Jun 2011, A E [Gmail] wrote:
>
>  Right! I did read that, the problem is how do I do this in bash?? I tried
>> read the result in and just post a Noop kind of a thing just to tell that I
>> read something, but it didn't help. I also explicitly did that in the perl
>> script, but doesn't work. It only works in PHP.
>>
>
> Bash would probably be my last choice of language to write an AGI with.
>
> Personally, I use C because it is my sharpest tool and because you can
> execute hundreds of AGIs written in C in the time it takes to load the Perl
> or PHP interpreter and parse your script.
>
> I suspect that the problems you are experiencing with Perl may have
> something to do with flushing STDOUT or reading the complete response from
> STDIN.
>
> I strongly suggest using an existing library for the language of your
> choice.
>
>
> Copy that. Not planning to write an AGI script in bash actually...it will
be written in C# running on a remote system. I was just doing a quick PoC to
figure out how would I use the stream file function to actually read audio
files over the network and even though I used to teach Perl 10+ yrs ago, I
don't do much scripting/coding for a long time, so the brain doesn't think
like a coder anymore. Just needed to try various tricks w.r.t to how can I
dynamically bring over audio files from another server, convert them to the
codec of my channel and then play/store them locally (cache if you will) but
wanted to learn the right way to do it with a local file first before I
tried something fancier. Guess I'll continue playing with the php script
that worked and once I figure the process out, will give it to the C# dev to
implement. Can't believe I wasted more than 2-3 hrs on this :(

BTW, I'd raised that a while ago, and got no conclusive response. How / what
is the best way to stream audio files (not MOH/Internet Radio/TV and what
not) inside a dialplan using AGI without comprising performance/adding
latency too much. no examples of shout/ICE I could find that show how to do
that simply by allowing me to run a web server remotely and use a shoutcast
module to play the audio right into the channel

ideas?
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