Re: [asterisk-users] PlayBack

[astuserlist]

Hi,
thanks, ControlPlayback is working.

I have # autoload = no
so I have to load by hand or put it in modules.conf


module load app_controlplayback.so


; 20221215
; ls -lah /usr/lib/asterisk/modules/app_controlplayback.so
; module show like play
; module load app_controlplayback.so







Am 14.12.22 um 13:08 schrieb Joshua C. Colp:
On Wed, Dec 14, 2022 at 8:02 AM > wrote:


Hi,

not installed
pbx.c:2907 pbx_extension_helper: No application 'ControlPlayback' for
extension

My version or Linux versionto old? I installed Asterisk using #aptitude
installed asterisk
Asterisk 16.2.1~dfsg-1+deb10u2 built by nobody @ buildd.debian.org
 on a
unknown running Linux on 2020-08-26 22:53:40 UTC


The module exists in 16.2.1, it is from the app_controlplayback module. 
The project doesn't build those packages, so I don't know how they were 
built. I also don't know your configuration (you might have modules.conf 
set to only load specific things, in which case you'd need to also load 
app_controlplayback).


--
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com  and 
www.asterisk.org 




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Re: [asterisk-users] PlayBack

[Joshua C. Colp]

On Wed, Dec 14, 2022 at 8:02 AM  wrote:

> Hi,
>
> not installed
> pbx.c:2907 pbx_extension_helper: No application 'ControlPlayback' for
> extension
>
> My version or Linux versionto old? I installed Asterisk using #aptitude
> installed asterisk
> Asterisk 16.2.1~dfsg-1+deb10u2 built by nobody @ buildd.debian.org on a
> unknown running Linux on 2020-08-26 22:53:40 UTC
>

The module exists in 16.2.1, it is from the app_controlplayback module. The
project doesn't build those packages, so I don't know how they were built.
I also don't know your configuration (you might have modules.conf set to
only load specific things, in which case you'd need to also load
app_controlplayback).

-- 
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] PlayBack

[astuserlist]

Hi,

not installed
pbx.c:2907 pbx_extension_helper: No application 'ControlPlayback' for 
extension


My version or Linux versionto old? I installed Asterisk using #aptitude 
installed asterisk
Asterisk 16.2.1~dfsg-1+deb10u2 built by nobody @ buildd.debian.org on a 
unknown running Linux on 2020-08-26 22:53:40 UTC


Bye Thomas



Am 14.12.22 um 12:34 schrieb Joshua C. Colp:
On Wed, Dec 14, 2022 at 7:32 AM > wrote:


Hi,
Iam using Playback do play an sound file.
Is there a programm where I can move forward an backward within the
sound file with DTMF tones.


The ControlPlayback dialplan application[1] allows this.

[1] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Application_ControlPlayback 
 



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Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com  and 
www.asterisk.org 




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Re: [asterisk-users] PlayBack

[Joshua C. Colp]

On Wed, Dec 14, 2022 at 7:32 AM  wrote:

> Hi,
> Iam using Playback do play an sound file.
> Is there a programm where I can move forward an backward within the
> sound file with DTMF tones.
>

The ControlPlayback dialplan application[1] allows this.

[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Application_ControlPlayback


-- 
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
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Re: [asterisk-users] Playback/background audio from MySQL BLOB

[A J Stiles]

On Tuesday 23 Sep 2014, Steve Edwards wrote:
 For some applications, storing recorded audio (prompts and caller
 recordings) as a BLOB in MySQL has advantages.
 
 So, once I have the audio in the database, how can I play it?
 
 Creating temporary files seems so tacky.
 
 Is there another way to playback or background audio either by specifying
 a URL or from a memory buffer (either C or PHP)?

Depending how many messages you have, you could use a named pipe  (FIFO)  or a 
Unix-domain socket for each one; and have the individual backend processes 
interrogate the database and dump the contents of the relevant field down it.  
As far as Asterisk is concerned, the socket / FIFO looks just like a file; it 
doesn't care much that the data in it is really coming from a process on the 
other end.  This obviously suffers from the problem of decreasing 
manageability, the more message files you have.

But personally, I'd just store the filenames in the database; and rely on the 
unix filesystem for storing the actual file contents.  After all, that's what a 
filesystem is for.

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] Playback/background audio from MySQL BLOB

[Hoggins!]

Hello,

Le 23/09/2014 23:38, Steve Edwards a écrit :
 On 09/23/2014 02:17 PM, Steve Edwards wrote:

 For some applications, storing recorded audio (prompts and caller
 recordings) as a BLOB in MySQL has advantages.

 On Tue, 23 Sep 2014, Don Kelly wrote:

 I'm curious about what the advantages are of storing audio in a blob.
 Wouldn't it be more efficient to store it in a file and just put the
 filename in the database?

 Multiple web servers, multiple Asterisk servers, multiple DB servers,
 synchronizing filesystems vs db, etc.

 It appears to eliminate some problems, but Asterisk limiting audio
 playback to files seems like a tough obstacle.

We solved this problem by storing the voicemail with IMAP. It is
possible to access IMAP simultaneously from different locations, and a
lot of implementations offer the ability to integrate IMAP into
web-applications.

That's how we did it anyway.

Best regards.




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Re: [asterisk-users] Playback/background audio from MySQL BLOB

[Chris Bagnall]

On 24/9/14 10:36 am, A J Stiles wrote:

But personally, I'd just store the filenames in the database; and rely on the
unix filesystem for storing the actual file contents.  After all, that's what a
filesystem is for.


This.

Shocking as it might appear, filesystems are remarkably good at storing 
files. They were designed to do it. Why try to shoehorn a database into 
doing something it wasn't designed to do (and isn't particularly good at 
doing)?


Kind regards,

Chris
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Re: [asterisk-users] Playback/background audio from MySQL BLOB

[Jeff LaCoursiere]

On 09/23/2014 10:53 PM, Don Kelly wrote:

On Tue, 23 Sep 2014, Steve Edwards wrote:


  On 09/23/2014 02:17 PM, Steve Edwards wrote:

  For some applications, storing recorded audio (prompts and caller
  recordings) as a BLOB in MySQL has advantages.

On Tue, 23 Sep 2014, Don Kelly wrote:


  I'm curious about what the advantages are of storing audio in a blob.
  Wouldn't it be more efficient to store it in a file and just put the
filename in the database?

Multiple web servers, multiple Asterisk servers, multiple DB servers,
synchronizing filesystems vs db, etc.

It appears to eliminate some problems, but Asterisk limiting audio
playback to files seems like a tough obstacle.



Mike said:
Maybe make the audio files available to all servers via a single NFS
directory?  Probably not a good solution if the servers aren't co-located.


Maybe someone could write a Linux device file that would return the blob's
content as a file read.




I beat you to that one ;)  That is exactly what a named pipe (fifo) is.  
Asterisk would read it like a sound file, and the AGI would dump the 
BLOB to it on demand.  It would work, but you can't have more than one 
process at a time reading from it, so that's a further complication...


j

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Re: [asterisk-users] Playback/background audio from MySQL BLOB

[Jeff LaCoursiere]

On 09/23/2014 02:17 PM, Steve Edwards wrote:
For some applications, storing recorded audio (prompts and caller 
recordings) as a BLOB in MySQL has advantages.


So, once I have the audio in the database, how can I play it?

Creating temporary files seems so tacky.

Is there another way to playback or background audio either by 
specifying a URL or from a memory buffer (either C or PHP)?




How about a named pipe (fifo)?  Of course then you might have issues 
with simultaneous calls.  You would have to have a pool of them and 
somehow manage locking them...


j

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Re: [asterisk-users] Playback/background audio from MySQL BLOB

[Don Kelly]

On 09/23/2014 02:17 PM, Steve Edwards wrote:
 For some applications, storing recorded audio (prompts and caller
 recordings) as a BLOB in MySQL has advantages.
Jeff sez:
How about a named pipe (fifo)?  Of course then you might have issues with
simultaneous calls.  You would have to have a pool of them and somehow
manage locking them...

J

I'm curious about what the advantages are of storing audio in a blob.
Wouldn't it be more efficient to store it in a file and just put the
filename in the database?

  --Don



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Re: [asterisk-users] Playback/background audio from MySQL BLOB

[Steve Edwards]

On 09/23/2014 02:17 PM, Steve Edwards wrote:



For some applications, storing recorded audio (prompts and caller
recordings) as a BLOB in MySQL has advantages.


On Tue, 23 Sep 2014, Don Kelly wrote:


I'm curious about what the advantages are of storing audio in a blob.
Wouldn't it be more efficient to store it in a file and just put the
filename in the database?


Multiple web servers, multiple Asterisk servers, multiple DB servers, 
synchronizing filesystems vs db, etc.


It appears to eliminate some problems, but Asterisk limiting audio 
playback to files seems like a tough obstacle.


--
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] Playback/background audio from MySQL BLOB

[Mike]

On Tue, 23 Sep 2014, Steve Edwards wrote:


 On 09/23/2014 02:17 PM, Steve Edwards wrote:



  For some applications, storing recorded audio (prompts and caller
  recordings) as a BLOB in MySQL has advantages.


On Tue, 23 Sep 2014, Don Kelly wrote:


 I'm curious about what the advantages are of storing audio in a blob.
 Wouldn't it be more efficient to store it in a file and just put the
 filename in the database?


Multiple web servers, multiple Asterisk servers, multiple DB servers, 
synchronizing filesystems vs db, etc.


It appears to eliminate some problems, but Asterisk limiting audio playback 
to files seems like a tough obstacle.


Maybe make the audio files available to all servers via a single NFS 
directory?  Probably not a good solution if the servers aren't co-located.


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Re: [asterisk-users] Playback/background audio from MySQL BLOB

[Don Kelly]

On Tue, 23 Sep 2014, Steve Edwards wrote:

  On 09/23/2014 02:17 PM, Steve Edwards wrote:

   For some applications, storing recorded audio (prompts and caller
   recordings) as a BLOB in MySQL has advantages.

 On Tue, 23 Sep 2014, Don Kelly wrote:

  I'm curious about what the advantages are of storing audio in a blob.
  Wouldn't it be more efficient to store it in a file and just put the  
 filename in the database?

 Multiple web servers, multiple Asterisk servers, multiple DB servers, 
 synchronizing filesystems vs db, etc.

 It appears to eliminate some problems, but Asterisk limiting audio 
 playback to files seems like a tough obstacle.



Mike said:
Maybe make the audio files available to all servers via a single NFS
directory?  Probably not a good solution if the servers aren't co-located.


Maybe someone could write a Linux device file that would return the blob's
content as a file read.

  --Don


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Re: [asterisk-users] Playback on h exten

[Leandro Dardini]

2013/2/21 Enrico Pasqualotto e.pasqualo...@netspin.it

 Hi all, I'm trying to setup a Quiz/feedback for caller of call center when
 a agent hangup.
 I use Asterisk 1.8.16 and I'm trying with Queue and Dial with options c
 and g but every time I try to play something I got:

 -- Executing [301@from-test:1] Dial(SIP/300-0045,
 SIP/301,60,rjtTg) in new stack
 -- Called SIP/301
 -- SIP/301-0046 is ringing
 -- SIP/301-0046 answered SIP/300-0045
 -- Auto fallthrough, channel 'SIP/300-0045' status is 'ANSWER'
 -- Executing [h@from-test:1] Goto(SIP/300-0045, play,s,1) in
 new stack
 -- Goto (play,s,1)
 -- Executing [s@play:1] NoOp(SIP/300-0045, play) in new stack
 -- Executing [s@play:2] SayDigits(SIP/300-0045, 123579) in
 new stack
 [Feb 21 10:35:00] WARNING[31945]: file.c:833 ast_readaudio_callback:
 Failed to write frame
 -- SIP/300-0045 Playing 'digits/1.ulaw' (language 'en')
   == Spawn extension (play, s, 2) exited non-zero on 'SIP/300-0045'

 This is my dialplan:

 [from-test]
 exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg)
 exten = h,1,Goto(play,s,1)

 [play]
 exten = s,1,Noop(play)
 exten = s,2,Saydigits(123579)


 Anyone can help me?

 Thanks

 Enrico.


If you choose to go with the Dial command and use the g option, you have
not to use the h extension, but just provide a next priority. Your
dialplan has to be:

[from-test]
exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg)
*exten = _X.,2,Goto(play,s,1)*

[play]
exten = s,1,Noop(play)
exten = s,2,Saydigits(123579)

Leandro
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Re: [asterisk-users] Playback on h exten

[Enrico Pasqualotto]

Yes, correct now it works for Dial. 
I think is the same with c option on Queue, do you think there's a way to do 
it on h exten? 
My goal is to inject my dialplan on hangup macro. 

Enrico. 
- Messaggio originale -

 If you choose to go with the Dial command and use the g option, you
 have not to use the h extension, but just provide a next priority.
 Your dialplan has to be:

 [from-test]
 exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg)
 exten = _X.,2,Goto(play,s,1)

 [play]
 exten = s,1,Noop(play)
 exten = s,2,Saydigits(123579)

 Leandro

-- 

-- 
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skype://epasqualotto :: http://www.linkedin.com/in/epasqualotto 
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Re: [asterisk-users] Playback on h exten

[A J Stiles]

On Thursday 21 February 2013, Enrico Pasqualotto wrote:
 Hi all, I'm trying to setup a Quiz/feedback for caller of call center when
 a agent hangup. I use Asterisk 1.8.16 and I'm trying with Queue and Dial
 with options c and g but every time I try to play something I got:
 .  stuff deleted .
 This is my dialplan:
 
 [from-test]
 exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg)
 exten = h,1,Goto(play,s,1)
 
 [play]
 exten = s,1,Noop(play)
 exten = s,2,Saydigits(123579)
 
 
 Anyone can help me?

By the time you reach extension h, it's too late for playing audio; the call 
has already been hung up, and the process of freeing up resources has begun.

If you are using the g modifier to the Dial() command then the call doesn't 
jump to extension h, but carries on with the next step if the far end hangs 
up.  So what you probably want is something more like this:

[from-test]
exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg)
exten = _X.,2,Noop(play)
exten = _X.,3,Saydigits(123579)


-- 
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Answers come *after* questions.

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Re: [asterisk-users] Playback on h exten

[Leandro Dardini]

The h exten is triggered when the channel is hangup, so you cannot send any
voice data on it.

Leandro

2013/2/21 Enrico Pasqualotto e.pasqualo...@netspin.it

 Yes, correct now it works for Dial.
 I think is the same with c option on Queue, do you think there's a way
 to do it on h exten?
 My goal is to inject my dialplan on hangup macro.

 Enrico.
 --


 If you choose to go with the Dial command and use the g option, you have
 not to use the h extension, but just provide a next priority. Your
 dialplan has to be:

 [from-test]
 exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg)
 *exten = _X.,2,Goto(play,s,1)*

 [play]
 exten = s,1,Noop(play)
 exten = s,2,Saydigits(123579)

 Leandro


 --
 --
 Pasqualotto Enrico
 cell. +39 3473292620
 skype://epasqualotto :: http://www.linkedin.com/in/epasqualotto
 http://www.netspin.it :: e.pasqualo...@netspin.it

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Re: [asterisk-users] Playback on h exten

[Bharat Lalcheta]

Any application with audio stream like playback or background will not work
in h priority, use of g in Dial() and c in Queue() is best approach for the
same.

Refer following link for h priority detail explanation
http://www.voip-info.org/wiki/view/Asterisk+h+extension

Regards,

Bharat
On Thu, Feb 21, 2013 at 3:53 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:

 On Thursday 21 February 2013, Enrico Pasqualotto wrote:
  Hi all, I'm trying to setup a Quiz/feedback for caller of call center
 when
  a agent hangup. I use Asterisk 1.8.16 and I'm trying with Queue and Dial
  with options c and g but every time I try to play something I got:
  .  stuff deleted .
  This is my dialplan:
 
  [from-test]
  exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg)
  exten = h,1,Goto(play,s,1)
 
  [play]
  exten = s,1,Noop(play)
  exten = s,2,Saydigits(123579)
 
 
  Anyone can help me?

 By the time you reach extension h, it's too late for playing audio; the
 call
 has already been hung up, and the process of freeing up resources has
 begun.

 If you are using the g modifier to the Dial() command then the call doesn't
 jump to extension h, but carries on with the next step if the far end hangs
 up.  So what you probably want is something more like this:

 [from-test]
 exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg)
 exten = _X.,2,Noop(play)
 exten = _X.,3,Saydigits(123579)


 --
 AJS

 Answers come *after* questions.

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-- 
Bharat Lalcheta
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Re: [asterisk-users] Playback with noanswer in AGI

[Zohair Raza]

Hi Sammy,

Thanks for input.

I have an eyebeam softphone registered with Asterisk 1.8.6 locally and from
agi, I pass this

$filetoplay = 'congestion';
$agi-exec(Progress);
$agi-exec(Playback $filetoplay,noanswer);

Have tried putting file in .gsm and .wav formats, I hear ringing tone
instead of playback

Please have a look at sip-trace

--- SIP read from UDP:176.249.0.50:8721 ---
INVITE sip:100@176.249.0.77 SIP/2.0
To: sip:100@176.249.0.77
From: Zohairsip:1000@176.249.0.77;tag=7f222672
Via: SIP/2.0/UDP 176.249.0.50:8721
;branch=z9hG4bK-d87543-521938753-1--d87543-;rport
Call-ID: 2932f90ef302332b
CSeq: 2 INVITE
Contact: sip:1000@176.249.0.50:8721
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 3006o stamp 17551
Authorization: Digest
username=1000,realm=asterisk,nonce=2abce759,uri=sip:100@176.249.0.77
,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5
Content-Length: 269

v=0
o=- 4333518 4333604 IN IP4 176.249.0.50
s=eyeBeam
c=IN IP4 176.249.0.50
t=0 0
m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101
a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506
a=fmtp:101 0-15
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
-
--- (13 headers 11 lines) ---
Sending to 176.249.0.50:8721 (no NAT)
sing INVITE request as basis request - 2932f90ef302332b
Found peer '1000' for '1000' from 176.249.0.50:8721
  == Using SIP RTP CoS mark 5
Found RTP audio format 100
Found RTP audio format 6
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 5
Found RTP audio format 101
Found audio description format speex for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e
(gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing),
combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 176.249.0.50:6506
Looking for 100 in default (domain 176.249.0.77)
list_route: hop: sip:1000@176.249.0.50:8721

--- Transmitting (no NAT) to 176.249.0.50:8721 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 176.249.0.50:8721
;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721
From: Zohairsip:1000@176.249.0.77;tag=7f222672
To: sip:100@176.249.0.77
Call-ID: 2932f90ef302332b
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: sip:100@176.249.0.77:5060
Content-Length: 0



-- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID)
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php
-- AGI Script Executing Application: (Progress) Options: ()
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

--- Transmitting (no NAT) to 176.249.0.50:8721 ---
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 176.249.0.50:8721
;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721
From: Zohairsip:1000@176.249.0.77;tag=7f222672
To: sip:100@176.249.0.77;tag=as01491743
Call-ID: 2932f90ef302332b
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: sip:100@176.249.0.77:5060
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 1225456982 1225456982 IN IP4 176.249.0.77
s=Asterisk PBX 1.8.0
c=IN IP4 176.249.0.77
t=0 0
m=audio 15918 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


-- AGI Script Executing Application: (Playback) Options:
(congestion,noanswer)
-- SIP/1000-0019 Playing 'congestion.slin' (language 'en')
-- SIP/1000-0019AGI Script agi.php completed, returning 0


Regards,
Zohair Raza


On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.com wrote:

 Hey Danny,

 I've this thing exactly running and working as Zohair mentioned! i.e I do
 not answer() the call rather put a progress() and soon after that playing
 back the sound file from playback with noanswer and then I get the file
 streaming as 183-Session progress file.

 I do understand that playing any sound file before establishing any audio
 session between two end point will result in no-adio from playback() BUT
 the combination of progress() and playback(,noanswer) works fine for me.

 What I think the issue could be for Zohair is that its requesting/incoming
 session(carrier) isn't allowing the 183-Session progress.

 Zohair can you do a SIP trace for this particular call along with the
 dialplan executing for it!?

 Regards,
 Sammy.


 On Tue, Feb 7, 2012 at 11:55 AM, Zohair Raza engineerzuhairr...@gmail.com
  wrote:

 

Re: [asterisk-users] Playback with noanswer in AGI

[Zohair Raza]

Sammy,

Problem is at phones, with a linksys phone it works but with eyebeam and
fanvill it doesn't

Maybe they don't support early media.

I think i will have to stick with ResetCDR and that will be okay now as
I've modified the code for that

Thank you

Regards,
Zohair Raza


On Tue, Feb 7, 2012 at 12:09 PM, Zohair Raza
engineerzuhairr...@gmail.comwrote:

 Hi Sammy,

 Thanks for input.

 I have an eyebeam softphone registered with Asterisk 1.8.6 locally and
 from agi, I pass this

 $filetoplay = 'congestion';
  $agi-exec(Progress);
 $agi-exec(Playback $filetoplay,noanswer);

 Have tried putting file in .gsm and .wav formats, I hear ringing tone
 instead of playback

 Please have a look at sip-trace

 --- SIP read from UDP:176.249.0.50:8721 ---
 INVITE sip:100@176.249.0.77 SIP/2.0
 To: sip:100@176.249.0.77
 From: Zohairsip:1000@176.249.0.77;tag=7f222672
 Via: SIP/2.0/UDP 176.249.0.50:8721
 ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport
 Call-ID: 2932f90ef302332b
 CSeq: 2 INVITE
 Contact: sip:1000@176.249.0.50:8721
 Max-Forwards: 70
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
 SUBSCRIBE, INFO
 Content-Type: application/sdp
 User-Agent: eyeBeam release 3006o stamp 17551
 Authorization: Digest
 username=1000,realm=asterisk,nonce=2abce759,uri=
 sip:100@176.249.0.77
 ,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5
 Content-Length: 269

 v=0
 o=- 4333518 4333604 IN IP4 176.249.0.50
 s=eyeBeam
 c=IN IP4 176.249.0.50
 t=0 0
 m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101
 a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506
 a=fmtp:101 0-15
 a=rtpmap:100 speex/16000
 a=rtpmap:101 telephone-event/8000
 a=sendrecv
 -
 --- (13 headers 11 lines) ---
 Sending to 176.249.0.50:8721 (no NAT)
 sing INVITE request as basis request - 2932f90ef302332b
 Found peer '1000' for '1000' from 176.249.0.50:8721
   == Using SIP RTP CoS mark 5
 Found RTP audio format 100
 Found RTP audio format 6
 Found RTP audio format 0
 Found RTP audio format 8
 Found RTP audio format 3
 Found RTP audio format 18
 Found RTP audio format 5
 Found RTP audio format 101
 Found audio description format speex for ID 100
 Found audio description format telephone-event for ID 101
 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e
 (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing),
 combined - 0xc (ulaw|alaw)
 Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
 (telephone-event|), combined - 0x1 (telephone-event|)
 Peer audio RTP is at port 176.249.0.50:6506
 Looking for 100 in default (domain 176.249.0.77)
 list_route: hop: sip:1000@176.249.0.50:8721

 --- Transmitting (no NAT) to 176.249.0.50:8721 ---
 SIP/2.0 100 Trying
 Via: SIP/2.0/UDP 176.249.0.50:8721
 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721
 From: Zohairsip:1000@176.249.0.77;tag=7f222672
 To: sip:100@176.249.0.77
 Call-ID: 2932f90ef302332b
 CSeq: 2 INVITE
 Server: Asterisk PBX 1.8.0
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
 PUBLISH
 Supported: replaces, timer
 Contact: sip:100@176.249.0.77:5060
 Content-Length: 0


 
 -- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID)
 -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php
 -- AGI Script Executing Application: (Progress) Options: ()
 Audio is at 5060
 Adding codec 0x4 (ulaw) to SDP
 Adding codec 0x8 (alaw) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP

 --- Transmitting (no NAT) to 176.249.0.50:8721 ---
 SIP/2.0 183 Session Progress
 Via: SIP/2.0/UDP 176.249.0.50:8721
 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721
 From: Zohairsip:1000@176.249.0.77;tag=7f222672
 To: sip:100@176.249.0.77;tag=as01491743
 Call-ID: 2932f90ef302332b
 CSeq: 2 INVITE
 Server: Asterisk PBX 1.8.0
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
 PUBLISH
 Supported: replaces, timer
 Contact: sip:100@176.249.0.77:5060
 Content-Type: application/sdp
 Content-Length: 258

 v=0
 o=root 1225456982 1225456982 IN IP4 176.249.0.77
 s=Asterisk PBX 1.8.0
 c=IN IP4 176.249.0.77
 t=0 0
 m=audio 15918 RTP/AVP 0 8 101
 a=rtpmap:0 PCMU/8000
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=ptime:20
 a=sendrecv

 
 -- AGI Script Executing Application: (Playback) Options:
 (congestion,noanswer)
 -- SIP/1000-0019 Playing 'congestion.slin' (language 'en')
 -- SIP/1000-0019AGI Script agi.php completed, returning 0


 Regards,
 Zohair Raza


 On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.com wrote:

 Hey Danny,

 I've this thing exactly running and working as Zohair mentioned! i.e I do
 not answer() the call rather put a progress() and soon after that playing
 back the sound file from playback with noanswer and then I get the file
 streaming as 183-Session progress file.

 I do understand that playing any sound file before establishing any audio
 session 

Re: [asterisk-users] Playback with noanswer in AGI

[Sammy Govind]

Hi,

Given invites seems fine, can you take a wireshark trace of the call on
your eyebeam machine! from that wireshark trace use telephony calls options
and hear if you are actually receiving RTPs on your system. If you could
hear the played back sound file on your eyembeam machine . this would mean
that your eyebeam client is not good enough to play media while its in 183
session progress.

Also can you send me the short sample php-agi script you are executing so i
actually test this on my virtual machines as well.

Regards,
Sammy

On Tue, Feb 7, 2012 at 1:09 PM, Zohair Raza engineerzuhairr...@gmail.comwrote:

 Hi Sammy,

 Thanks for input.

 I have an eyebeam softphone registered with Asterisk 1.8.6 locally and
 from agi, I pass this

 $filetoplay = 'congestion';
 $agi-exec(Progress);
 $agi-exec(Playback $filetoplay,noanswer);

 Have tried putting file in .gsm and .wav formats, I hear ringing tone
 instead of playback

 Please have a look at sip-trace

 --- SIP read from UDP:176.249.0.50:8721 ---
 INVITE sip:100@176.249.0.77 SIP/2.0
 To: sip:100@176.249.0.77
 From: Zohairsip:1000@176.249.0.77;tag=7f222672
 Via: SIP/2.0/UDP 176.249.0.50:8721
 ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport
 Call-ID: 2932f90ef302332b
 CSeq: 2 INVITE
 Contact: sip:1000@176.249.0.50:8721
 Max-Forwards: 70
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
 SUBSCRIBE, INFO
 Content-Type: application/sdp
 User-Agent: eyeBeam release 3006o stamp 17551
 Authorization: Digest
 username=1000,realm=asterisk,nonce=2abce759,uri=
 sip:100@176.249.0.77
 ,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5
 Content-Length: 269

 v=0
 o=- 4333518 4333604 IN IP4 176.249.0.50
 s=eyeBeam
 c=IN IP4 176.249.0.50
 t=0 0
 m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101
 a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506
 a=fmtp:101 0-15
 a=rtpmap:100 speex/16000
 a=rtpmap:101 telephone-event/8000
 a=sendrecv
 -
 --- (13 headers 11 lines) ---
 Sending to 176.249.0.50:8721 (no NAT)
 sing INVITE request as basis request - 2932f90ef302332b
 Found peer '1000' for '1000' from 176.249.0.50:8721
   == Using SIP RTP CoS mark 5
 Found RTP audio format 100
 Found RTP audio format 6
 Found RTP audio format 0
 Found RTP audio format 8
 Found RTP audio format 3
 Found RTP audio format 18
 Found RTP audio format 5
 Found RTP audio format 101
 Found audio description format speex for ID 100
 Found audio description format telephone-event for ID 101
 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e
 (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing),
 combined - 0xc (ulaw|alaw)
 Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
 (telephone-event|), combined - 0x1 (telephone-event|)
 Peer audio RTP is at port 176.249.0.50:6506
 Looking for 100 in default (domain 176.249.0.77)
 list_route: hop: sip:1000@176.249.0.50:8721

 --- Transmitting (no NAT) to 176.249.0.50:8721 ---
 SIP/2.0 100 Trying
 Via: SIP/2.0/UDP 176.249.0.50:8721
 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721
 From: Zohairsip:1000@176.249.0.77;tag=7f222672
 To: sip:100@176.249.0.77
 Call-ID: 2932f90ef302332b
 CSeq: 2 INVITE
 Server: Asterisk PBX 1.8.0
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
 PUBLISH
 Supported: replaces, timer
 Contact: sip:100@176.249.0.77:5060
 Content-Length: 0


 
 -- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID)
 -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php
 -- AGI Script Executing Application: (Progress) Options: ()
 Audio is at 5060
 Adding codec 0x4 (ulaw) to SDP
 Adding codec 0x8 (alaw) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP

 --- Transmitting (no NAT) to 176.249.0.50:8721 ---
 SIP/2.0 183 Session Progress
 Via: SIP/2.0/UDP 176.249.0.50:8721
 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721
 From: Zohairsip:1000@176.249.0.77;tag=7f222672
 To: sip:100@176.249.0.77;tag=as01491743
 Call-ID: 2932f90ef302332b
 CSeq: 2 INVITE
 Server: Asterisk PBX 1.8.0
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
 PUBLISH
 Supported: replaces, timer
 Contact: sip:100@176.249.0.77:5060
 Content-Type: application/sdp
 Content-Length: 258

 v=0
 o=root 1225456982 1225456982 IN IP4 176.249.0.77
 s=Asterisk PBX 1.8.0
 c=IN IP4 176.249.0.77
 t=0 0
 m=audio 15918 RTP/AVP 0 8 101
 a=rtpmap:0 PCMU/8000
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=ptime:20
 a=sendrecv

 
 -- AGI Script Executing Application: (Playback) Options:
 (congestion,noanswer)
 -- SIP/1000-0019 Playing 'congestion.slin' (language 'en')
 -- SIP/1000-0019AGI Script agi.php completed, returning 0


 Regards,
 Zohair Raza


 On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.com wrote:

 Hey Danny,

 I've this thing exactly running and working as Zohair mentioned! i.e I do
 not answer() the 

Re: [asterisk-users] Playback with noanswer in AGI

[Sammy Govind]

Exactly that's what I expected.
Great - now have fun

On Tue, Feb 7, 2012 at 2:09 PM, Zohair Raza engineerzuhairr...@gmail.comwrote:

 Sammy,

 Problem is at phones, with a linksys phone it works but with eyebeam and
 fanvill it doesn't

 Maybe they don't support early media.

 I think i will have to stick with ResetCDR and that will be okay now as
 I've modified the code for that

 Thank you

 Regards,
 Zohair Raza


 On Tue, Feb 7, 2012 at 12:09 PM, Zohair Raza engineerzuhairr...@gmail.com
  wrote:

 Hi Sammy,

 Thanks for input.

 I have an eyebeam softphone registered with Asterisk 1.8.6 locally and
 from agi, I pass this

 $filetoplay = 'congestion';
  $agi-exec(Progress);
 $agi-exec(Playback $filetoplay,noanswer);

 Have tried putting file in .gsm and .wav formats, I hear ringing tone
 instead of playback

 Please have a look at sip-trace

 --- SIP read from UDP:176.249.0.50:8721 ---
 INVITE sip:100@176.249.0.77 SIP/2.0
 To: sip:100@176.249.0.77
 From: Zohairsip:1000@176.249.0.77;tag=7f222672
 Via: SIP/2.0/UDP 176.249.0.50:8721
 ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport
 Call-ID: 2932f90ef302332b
 CSeq: 2 INVITE
 Contact: sip:1000@176.249.0.50:8721
 Max-Forwards: 70
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
 SUBSCRIBE, INFO
 Content-Type: application/sdp
 User-Agent: eyeBeam release 3006o stamp 17551
 Authorization: Digest
 username=1000,realm=asterisk,nonce=2abce759,uri=
 sip:100@176.249.0.77
 ,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5
 Content-Length: 269

 v=0
 o=- 4333518 4333604 IN IP4 176.249.0.50
 s=eyeBeam
 c=IN IP4 176.249.0.50
 t=0 0
 m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101
 a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506
 a=fmtp:101 0-15
 a=rtpmap:100 speex/16000
 a=rtpmap:101 telephone-event/8000
 a=sendrecv
 -
 --- (13 headers 11 lines) ---
 Sending to 176.249.0.50:8721 (no NAT)
 sing INVITE request as basis request - 2932f90ef302332b
 Found peer '1000' for '1000' from 176.249.0.50:8721
   == Using SIP RTP CoS mark 5
 Found RTP audio format 100
 Found RTP audio format 6
 Found RTP audio format 0
 Found RTP audio format 8
 Found RTP audio format 3
 Found RTP audio format 18
 Found RTP audio format 5
 Found RTP audio format 101
 Found audio description format speex for ID 100
 Found audio description format telephone-event for ID 101
 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e
 (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing),
 combined - 0xc (ulaw|alaw)
 Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
 (telephone-event|), combined - 0x1 (telephone-event|)
 Peer audio RTP is at port 176.249.0.50:6506
 Looking for 100 in default (domain 176.249.0.77)
 list_route: hop: sip:1000@176.249.0.50:8721

 --- Transmitting (no NAT) to 176.249.0.50:8721 ---
 SIP/2.0 100 Trying
 Via: SIP/2.0/UDP 176.249.0.50:8721
 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721
 From: Zohairsip:1000@176.249.0.77;tag=7f222672
 To: sip:100@176.249.0.77
 Call-ID: 2932f90ef302332b
 CSeq: 2 INVITE
 Server: Asterisk PBX 1.8.0
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
 PUBLISH
 Supported: replaces, timer
 Contact: sip:100@176.249.0.77:5060
 Content-Length: 0


 
 -- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID)
 -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php
 -- AGI Script Executing Application: (Progress) Options: ()
 Audio is at 5060
 Adding codec 0x4 (ulaw) to SDP
 Adding codec 0x8 (alaw) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP

 --- Transmitting (no NAT) to 176.249.0.50:8721 ---
 SIP/2.0 183 Session Progress
 Via: SIP/2.0/UDP 176.249.0.50:8721
 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721
 From: Zohairsip:1000@176.249.0.77;tag=7f222672
 To: sip:100@176.249.0.77;tag=as01491743
 Call-ID: 2932f90ef302332b
 CSeq: 2 INVITE
 Server: Asterisk PBX 1.8.0
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
 PUBLISH
 Supported: replaces, timer
 Contact: sip:100@176.249.0.77:5060
 Content-Type: application/sdp
 Content-Length: 258

 v=0
 o=root 1225456982 1225456982 IN IP4 176.249.0.77
 s=Asterisk PBX 1.8.0
 c=IN IP4 176.249.0.77
 t=0 0
 m=audio 15918 RTP/AVP 0 8 101
 a=rtpmap:0 PCMU/8000
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=ptime:20
 a=sendrecv

 
 -- AGI Script Executing Application: (Playback) Options:
 (congestion,noanswer)
 -- SIP/1000-0019 Playing 'congestion.slin' (language 'en')
 -- SIP/1000-0019AGI Script agi.php completed, returning 0


 Regards,
 Zohair Raza


 On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.com wrote:

 Hey Danny,

 I've this thing exactly running and working as Zohair mentioned! i.e I
 do not answer() the call rather put a progress() and soon after that
 playing back the sound file from playback with noanswer and then 

Re: [asterisk-users] Playback with noanswer in AGI

[Zohair Raza]

Yes,

Thanks


Regards,
Zohair Raza

On Tue, Feb 7, 2012 at 1:37 PM, Sammy Govind govoi...@gmail.com wrote:

 Exactly that's what I expected.
 Great - now have fun


 On Tue, Feb 7, 2012 at 2:09 PM, Zohair Raza 
 engineerzuhairr...@gmail.comwrote:

 Sammy,

 Problem is at phones, with a linksys phone it works but with eyebeam and
 fanvill it doesn't

 Maybe they don't support early media.

 I think i will have to stick with ResetCDR and that will be okay now as
 I've modified the code for that

 Thank you

 Regards,
 Zohair Raza


 On Tue, Feb 7, 2012 at 12:09 PM, Zohair Raza 
 engineerzuhairr...@gmail.com wrote:

 Hi Sammy,

 Thanks for input.

 I have an eyebeam softphone registered with Asterisk 1.8.6 locally and
 from agi, I pass this

 $filetoplay = 'congestion';
  $agi-exec(Progress);
 $agi-exec(Playback $filetoplay,noanswer);

 Have tried putting file in .gsm and .wav formats, I hear ringing tone
 instead of playback

 Please have a look at sip-trace

 --- SIP read from UDP:176.249.0.50:8721 ---
 INVITE sip:100@176.249.0.77 SIP/2.0
 To: sip:100@176.249.0.77
 From: Zohairsip:1000@176.249.0.77;tag=7f222672
 Via: SIP/2.0/UDP 176.249.0.50:8721
 ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport
 Call-ID: 2932f90ef302332b
 CSeq: 2 INVITE
 Contact: sip:1000@176.249.0.50:8721
 Max-Forwards: 70
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
 SUBSCRIBE, INFO
 Content-Type: application/sdp
 User-Agent: eyeBeam release 3006o stamp 17551
 Authorization: Digest
 username=1000,realm=asterisk,nonce=2abce759,uri=
 sip:100@176.249.0.77
 ,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5
 Content-Length: 269

 v=0
 o=- 4333518 4333604 IN IP4 176.249.0.50
 s=eyeBeam
 c=IN IP4 176.249.0.50
 t=0 0
 m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101
 a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506
 a=fmtp:101 0-15
 a=rtpmap:100 speex/16000
 a=rtpmap:101 telephone-event/8000
 a=sendrecv
 -
 --- (13 headers 11 lines) ---
 Sending to 176.249.0.50:8721 (no NAT)
 sing INVITE request as basis request - 2932f90ef302332b
 Found peer '1000' for '1000' from 176.249.0.50:8721
   == Using SIP RTP CoS mark 5
 Found RTP audio format 100
 Found RTP audio format 6
 Found RTP audio format 0
 Found RTP audio format 8
 Found RTP audio format 3
 Found RTP audio format 18
 Found RTP audio format 5
 Found RTP audio format 101
 Found audio description format speex for ID 100
 Found audio description format telephone-event for ID 101
 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e
 (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing),
 combined - 0xc (ulaw|alaw)
 Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
 (telephone-event|), combined - 0x1 (telephone-event|)
 Peer audio RTP is at port 176.249.0.50:6506
 Looking for 100 in default (domain 176.249.0.77)
 list_route: hop: sip:1000@176.249.0.50:8721

 --- Transmitting (no NAT) to 176.249.0.50:8721 ---
 SIP/2.0 100 Trying
 Via: SIP/2.0/UDP 176.249.0.50:8721
 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721
 From: Zohairsip:1000@176.249.0.77;tag=7f222672
 To: sip:100@176.249.0.77
 Call-ID: 2932f90ef302332b
 CSeq: 2 INVITE
 Server: Asterisk PBX 1.8.0
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
 INFO, PUBLISH
 Supported: replaces, timer
 Contact: sip:100@176.249.0.77:5060
 Content-Length: 0


 
 -- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID)
 -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php
 -- AGI Script Executing Application: (Progress) Options: ()
 Audio is at 5060
 Adding codec 0x4 (ulaw) to SDP
 Adding codec 0x8 (alaw) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP

 --- Transmitting (no NAT) to 176.249.0.50:8721 ---
 SIP/2.0 183 Session Progress
 Via: SIP/2.0/UDP 176.249.0.50:8721
 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721
 From: Zohairsip:1000@176.249.0.77;tag=7f222672
 To: sip:100@176.249.0.77;tag=as01491743
 Call-ID: 2932f90ef302332b
 CSeq: 2 INVITE
 Server: Asterisk PBX 1.8.0
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
 INFO, PUBLISH
 Supported: replaces, timer
 Contact: sip:100@176.249.0.77:5060
 Content-Type: application/sdp
 Content-Length: 258

 v=0
 o=root 1225456982 1225456982 IN IP4 176.249.0.77
 s=Asterisk PBX 1.8.0
 c=IN IP4 176.249.0.77
 t=0 0
 m=audio 15918 RTP/AVP 0 8 101
 a=rtpmap:0 PCMU/8000
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=ptime:20
 a=sendrecv

 
 -- AGI Script Executing Application: (Playback) Options:
 (congestion,noanswer)
 -- SIP/1000-0019 Playing 'congestion.slin' (language 'en')
 -- SIP/1000-0019AGI Script agi.php completed, returning 0


 Regards,
 Zohair Raza


 On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.comwrote:

 Hey Danny,

 I've this thing exactly running and working as Zohair mentioned! i.e I
 do not answer() the 

Re: [asterisk-users] Playback with noanswer in AGI

[Zohair Raza]

Confirmed as well, played back with wireshark and audio was there but phone
was ringing.

Thanks again.

Regards,
Zohair Raza

On Tue, Feb 7, 2012 at 1:37 PM, Sammy Govind govoi...@gmail.com wrote:

 Hi,

 Given invites seems fine, can you take a wireshark trace of the call on
 your eyebeam machine! from that wireshark trace use telephony calls options
 and hear if you are actually receiving RTPs on your system. If you could
 hear the played back sound file on your eyembeam machine . this would mean
 that your eyebeam client is not good enough to play media while its in 183
 session progress.

 Also can you send me the short sample php-agi script you are executing so
 i actually test this on my virtual machines as well.

 Regards,
 Sammy

 On Tue, Feb 7, 2012 at 1:09 PM, Zohair Raza 
 engineerzuhairr...@gmail.comwrote:

 Hi Sammy,

 Thanks for input.

 I have an eyebeam softphone registered with Asterisk 1.8.6 locally and
 from agi, I pass this

 $filetoplay = 'congestion';
  $agi-exec(Progress);
 $agi-exec(Playback $filetoplay,noanswer);

 Have tried putting file in .gsm and .wav formats, I hear ringing tone
 instead of playback

 Please have a look at sip-trace

 --- SIP read from UDP:176.249.0.50:8721 ---
 INVITE sip:100@176.249.0.77 SIP/2.0
 To: sip:100@176.249.0.77
 From: Zohairsip:1000@176.249.0.77;tag=7f222672
 Via: SIP/2.0/UDP 176.249.0.50:8721
 ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport
 Call-ID: 2932f90ef302332b
 CSeq: 2 INVITE
 Contact: sip:1000@176.249.0.50:8721
 Max-Forwards: 70
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
 SUBSCRIBE, INFO
 Content-Type: application/sdp
 User-Agent: eyeBeam release 3006o stamp 17551
 Authorization: Digest
 username=1000,realm=asterisk,nonce=2abce759,uri=
 sip:100@176.249.0.77
 ,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5
 Content-Length: 269

 v=0
 o=- 4333518 4333604 IN IP4 176.249.0.50
 s=eyeBeam
 c=IN IP4 176.249.0.50
 t=0 0
 m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101
 a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506
 a=fmtp:101 0-15
 a=rtpmap:100 speex/16000
 a=rtpmap:101 telephone-event/8000
 a=sendrecv
 -
 --- (13 headers 11 lines) ---
 Sending to 176.249.0.50:8721 (no NAT)
 sing INVITE request as basis request - 2932f90ef302332b
 Found peer '1000' for '1000' from 176.249.0.50:8721
   == Using SIP RTP CoS mark 5
 Found RTP audio format 100
 Found RTP audio format 6
 Found RTP audio format 0
 Found RTP audio format 8
 Found RTP audio format 3
 Found RTP audio format 18
 Found RTP audio format 5
 Found RTP audio format 101
 Found audio description format speex for ID 100
 Found audio description format telephone-event for ID 101
 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e
 (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing),
 combined - 0xc (ulaw|alaw)
 Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
 (telephone-event|), combined - 0x1 (telephone-event|)
 Peer audio RTP is at port 176.249.0.50:6506
 Looking for 100 in default (domain 176.249.0.77)
 list_route: hop: sip:1000@176.249.0.50:8721

 --- Transmitting (no NAT) to 176.249.0.50:8721 ---
 SIP/2.0 100 Trying
 Via: SIP/2.0/UDP 176.249.0.50:8721
 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721
 From: Zohairsip:1000@176.249.0.77;tag=7f222672
 To: sip:100@176.249.0.77
 Call-ID: 2932f90ef302332b
 CSeq: 2 INVITE
 Server: Asterisk PBX 1.8.0
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
 PUBLISH
 Supported: replaces, timer
 Contact: sip:100@176.249.0.77:5060
 Content-Length: 0


 
 -- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID)
 -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php
 -- AGI Script Executing Application: (Progress) Options: ()
 Audio is at 5060
 Adding codec 0x4 (ulaw) to SDP
 Adding codec 0x8 (alaw) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP

 --- Transmitting (no NAT) to 176.249.0.50:8721 ---
 SIP/2.0 183 Session Progress
 Via: SIP/2.0/UDP 176.249.0.50:8721
 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721
 From: Zohairsip:1000@176.249.0.77;tag=7f222672
 To: sip:100@176.249.0.77;tag=as01491743
 Call-ID: 2932f90ef302332b
 CSeq: 2 INVITE
 Server: Asterisk PBX 1.8.0
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
 PUBLISH
 Supported: replaces, timer
 Contact: sip:100@176.249.0.77:5060
 Content-Type: application/sdp
 Content-Length: 258

 v=0
 o=root 1225456982 1225456982 IN IP4 176.249.0.77
 s=Asterisk PBX 1.8.0
 c=IN IP4 176.249.0.77
 t=0 0
 m=audio 15918 RTP/AVP 0 8 101
 a=rtpmap:0 PCMU/8000
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=ptime:20
 a=sendrecv

 
 -- AGI Script Executing Application: (Playback) Options:
 (congestion,noanswer)
 -- SIP/1000-0019 Playing 'congestion.slin' (language 'en')
 -- SIP/1000-0019AGI Script agi.php completed, 

Re: [asterisk-users] Playback with noanswer in AGI

[Danny Nicholas]

You are mis-understanding the concept - the noanswer option is playing the
file as you requested, but since you aren't answering the call, no channel
is established to actually present the sound to you.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zohair Raza
Sent: Monday, February 06, 2012 12:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Playback with noanswer in AGI

 

Hi All, 

 

I want to play a file in agi but dont want to answer the call

 

I am dialing through sip phone and running asterisk 1.8.6,

 

I tried following with no luck

 

$agi-exec(Progress);

$agi-exec(Playback $filetoplay,noanswer);

$agi-hangup();

 

When I dial I can't hear the audio but if I answer the call or remove
noanswer argument I can hear the audio.

 

phpAGI's stream_file didn't help either. 

 

I ended up with ResetCDR() before hangup to reset billsec, duration and
disposition but don't want to do it this way.

 

What could be the problem?

 

From Voip-info.org :

noanswer: Play the sound file, but don't answer the channel first (if hasn't
been answered already). Not all channels support playing messages while
still on hook.

 

Is it because the channel is not supported?

 

 

Regards,

Zohair Raza

 

 

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Re: [asterisk-users] Playback with noanswer in AGI

[Zohair Raza]

Thanks for this explanation Dany!

Regards,
Zohair Raza


On Mon, Feb 6, 2012 at 10:11 PM, Danny Nicholas da...@debsinc.com wrote:

 You are mis-understanding the concept – the noanswer option is playing the
 file as you requested, but since you aren’t answering the call, no channel
 is established to actually present the sound to you.

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zohair Raza
 *Sent:* Monday, February 06, 2012 12:06 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] Playback with noanswer in AGI

 ** **

 Hi All, 

 ** **

 I want to play a file in agi but dont want to answer the call

 ** **

 I am dialing through sip phone and running asterisk 1.8.6,

 ** **

 I tried following with no luck

 ** **

 $agi-exec(Progress);

 $agi-exec(Playback $filetoplay,noanswer);

 $agi-hangup();

 ** **

 When I dial I can't hear the audio but if I answer the call or remove
 noanswer argument I can hear the audio.

 ** **

 phpAGI's stream_file didn't help either. 

 ** **

 I ended up with ResetCDR() before hangup to reset billsec, duration and
 disposition but don't want to do it this way.

 ** **

 What could be the problem?

 ** **

 From Voip-info.org :

 *noanswer*: Play the sound file, but don't answer the channel first (if
 hasn't been answered already). Not all channels support playing messages
 while still on hook.

 ** **

 Is it because the channel is not supported?

 ** **

 ** **

 Regards,

 Zohair Raza

 ** **

 ** **

 --
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 asterisk-users mailing list
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Re: [asterisk-users] Playback with noanswer in AGI

[Sammy Govind]

Hey Danny,

I've this thing exactly running and working as Zohair mentioned! i.e I do
not answer() the call rather put a progress() and soon after that playing
back the sound file from playback with noanswer and then I get the file
streaming as 183-Session progress file.

I do understand that playing any sound file before establishing any audio
session between two end point will result in no-adio from playback() BUT
the combination of progress() and playback(,noanswer) works fine for me.

What I think the issue could be for Zohair is that its requesting/incoming
session(carrier) isn't allowing the 183-Session progress.

Zohair can you do a SIP trace for this particular call along with the
dialplan executing for it!?

Regards,
Sammy.

On Tue, Feb 7, 2012 at 11:55 AM, Zohair Raza
engineerzuhairr...@gmail.comwrote:

 Thanks for this explanation Dany!

 Regards,
 Zohair Raza


 On Mon, Feb 6, 2012 at 10:11 PM, Danny Nicholas da...@debsinc.com wrote:

 You are mis-understanding the concept – the noanswer option is playing
 the file as you requested, but since you aren’t answering the call, no
 channel is established to actually present the sound to you.

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zohair Raza
 *Sent:* Monday, February 06, 2012 12:06 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] Playback with noanswer in AGI

 ** **

 Hi All, 

 ** **

 I want to play a file in agi but dont want to answer the call

 ** **

 I am dialing through sip phone and running asterisk 1.8.6,

 ** **

 I tried following with no luck

 ** **

 $agi-exec(Progress);

 $agi-exec(Playback $filetoplay,noanswer);

 $agi-hangup();

 ** **

 When I dial I can't hear the audio but if I answer the call or remove
 noanswer argument I can hear the audio.

 ** **

 phpAGI's stream_file didn't help either. 

 ** **

 I ended up with ResetCDR() before hangup to reset billsec, duration and
 disposition but don't want to do it this way.

 ** **

 What could be the problem?

 ** **

 From Voip-info.org :

 *noanswer*: Play the sound file, but don't answer the channel first (if
 hasn't been answered already). Not all channels support playing messages
 while still on hook.

 ** **

 Is it because the channel is not supported?

 ** **

 ** **

 Regards,

 Zohair Raza

 ** **

 ** **

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Re: [asterisk-users] Playback while dialing out

[Faisal Hanif]

Well as far as I know asterisk you can't play anything while channel is in
dialing state but music-on-hold. A solution to your problem is realtime
music-on-hold.

Following are possible steps,

1-Configure your asterisk for realtime music-on-hold
(http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf) so it
will get all mog class info from DB in realtime.
2-Before dialing a call create a moh class in db by hitting a query and
associate your target voice.mp3 files with that class.
3-Dial the call and associate that moh class using parameter.

Regards,

Faisal Hanif

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Boykin
Sent: Saturday, August 20, 2011 10:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Playback while dialing out

I am not sure you even read my mail, no music on hold option - it should
work dynamically with any file.

On Fri, Aug 19, 2011 at 6:18 PM, bakko asannu...@gmail.com wrote:
 Hi,

 you can configure a new music on hold, example:

 nano /etc/asterisk/musiconhold.conf

 [default1]
 mode=files
 directory=moh1

 and put the audio file in this directory; then change your dialplan like:

 exten = 500,1,NoOp
 exten = 500,2,Dial(SIP/14085551234@myprovider,m(default1))
 exten = 503,3,Hangup

 Regards

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Re: [asterisk-users] Playback while dialing out

[Jim Boykin]

A(x) does not accomplish this. It completes the playback and then
dials. What I would like is that dialing should start in parallel and
playback should stop as soon as early media or ringing starts.

Similarly, music-on-hold is not an option, it's too hard coded, I like
to be able to change playback file dynamically.

Any hints??



On Fri, Aug 19, 2011 at 7:14 AM, Eric Wieling ewiel...@nyigc.com wrote:
 Take a look at the A(x) and m options to dial.  In the Asterisk CLI core 
 show application dial for a the docs to Dial().

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Boykin
 Sent: Thursday, August 18, 2011 9:12 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Playback while dialing out

 Hi, please help me with dialplan below.

 My current dialplan looks like this, it plays a file and then connects the 
 caller to my phone by dialing out. As you can see, it waits for file to be 
 played completely before dialing out. What I would really like is that it 
 should play the file (preferably repetitively) and simultaneously dial out 
 the number, playback should stop as soon as dial answers or early media 
 detected.

 exten = 500,1,Answer
 exten = 500,2,Playback(wait-while-we-connect-you)
 exten = 500,3,Dial(SIP/14085551234@myprovider)

 How do I make it work?

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Re: [asterisk-users] Playback while dialing out

[bakko]

Hi,

you can configure a new music on hold, example:

nano /etc/asterisk/musiconhold.conf

[default1]
mode=files
directory=moh1

and put the audio file in this directory; then change your dialplan like:

exten = 500,1,NoOp
exten = 500,2,Dial(SIP/14085551234@myprovider,m(default1))
exten = 503,3,Hangup

Regards

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Re: [asterisk-users] Playback while dialing out

[Jim Boykin]

I am not sure you even read my mail, no music on hold option - it
should work dynamically with any file.

On Fri, Aug 19, 2011 at 6:18 PM, bakko asannu...@gmail.com wrote:
 Hi,

 you can configure a new music on hold, example:

 nano /etc/asterisk/musiconhold.conf

 [default1]
 mode=files
 directory=moh1

 and put the audio file in this directory; then change your dialplan like:

 exten = 500,1,NoOp
 exten = 500,2,Dial(SIP/14085551234@myprovider,m(default1))
 exten = 503,3,Hangup

 Regards

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Re: [asterisk-users] Playback while dialing out

[Eric Wieling]

Take a look at the A(x) and m options to dial.  In the Asterisk CLI core show 
application dial for a the docs to Dial(). 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Boykin
Sent: Thursday, August 18, 2011 9:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Playback while dialing out

Hi, please help me with dialplan below.

My current dialplan looks like this, it plays a file and then connects the 
caller to my phone by dialing out. As you can see, it waits for file to be 
played completely before dialing out. What I would really like is that it 
should play the file (preferably repetitively) and simultaneously dial out the 
number, playback should stop as soon as dial answers or early media detected.

exten = 500,1,Answer
exten = 500,2,Playback(wait-while-we-connect-you)
exten = 500,3,Dial(SIP/14085551234@myprovider)

How do I make it work?

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Re: [asterisk-users] Playback in uplink and recording in downlink

[Paul Belanger]

On 11-02-01 04:02 AM, Felix Dong wrote:
 I got a question to asterisk 1.6. Is it possible to playback a Audiofile in
 uplink and to record the downlink channel in another Audifile at the same
 time?
 
Yes, look at MixMonitor.

*CLI core show application MixMonitor

-- 
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Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Playback in uplink and recording in downlink

[Ruddy Gbaguidi]

Yes, you can use the Mixmonitor command.
But if you want to have only one party on the recording, you should use the
Monitor command without the 'm' option.


http://www.astblog.com/2011/02/01/asterisk-mixmonitor-cmd/


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: 2011-02-01 09:41 
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Playback in uplink and recording in downlink

On 11-02-01 04:02 AM, Felix Dong wrote:
 I got a question to asterisk 1.6. Is it possible to playback a 
 Audiofile in uplink and to record the downlink channel in another 
 Audifile at the same time?
 
Yes, look at MixMonitor.

*CLI core show application MixMonitor

--
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Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at:
http://digium.com  http://asterisk.org

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Re: [asterisk-users] Playback in the middle of a call though AMI

[Godson Gera]

Hi,

I am not using 1.6 but in 1.2 or 1.4 there is no straight forward way to do
this. The workaround i use it to pull the caller into conference and play
what ever I want using a agi script connected to the same conference room.

On Wed, Oct 20, 2010 at 4:35 PM, Gustavo Garcia Bernardo g...@tid.es wrote:

  Hi folks,



 Is it possible (asterisk 1.6) to trigger the playback of an audio file in
 the middle of a call using the Manager Interface?

 I’m looking for something like AMI PlayDTMF command but for audio files.



 Thanks a lot,

 G.





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Consultanthttp://blog.godson.in/2010/10/asterisk-vs-freeswitch-channel-tracking.html
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Re: [asterisk-users] Playback in the middle of a call though AMI

[Danny Nicholas]

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gustavo Garcia
Bernardo
Sent: Wednesday, October 20, 2010 6:06 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Playback in the middle of a call though AMI

 

Hi folks,

 

Is it possible (asterisk 1.6) to trigger the playback of an audio file in
the middle of a call using the Manager Interface?

I'm looking for something like AMI PlayDTMF command but for audio files.

 

Thanks a lot,

G.

 

I  don't use 1.6, but you might be able to do a command/playback to play the
file.

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Re: [asterisk-users] Playback during call

[Gabriel Ortiz Lour]

Thanks a lot, but my problem was the asterisk version!

The whisper mode on the ChanSpy of the 1.4.26 version is broken!

Upgraded version and it worked OK



2010/8/10 Jim Dickenson dicken...@cfmc.com

 Your ami packet is not setting the w option for chanspy, nor I am sure you
 can do this.

 You might want to create an additional exten that takes a variable from
 your ami packet and does the chanspy that way.

 I use an ami packet like this with extension that do the work.

 Action: Originate
 Channel: Local/do_playb...@cfmc_cdi_private
 Exten: do_chanspy
 Context: cfmc_cdi_private
 Priority: 1
 Variable: CfMC_ActionID=PlayBack
 Variable: CfMC_WhatToPlay=lyrics-louie-louie
 Variable: CfMC_WhoHear=SIP/GXP280_18-0002
 ActionID: PlayBack
 Async: true


 exten = do_playback,1,Answer()
 exten = do_playback,n,UserEvent(BeforePlayBack,ActionID:${CfMC_ActionID} 
 ${UNIQUEID}  ${CHANNEL}  ${CfMC_WhatToPlay}  ${CfMC_WhoHear})
 exten = do_playback,n,Wait(0.3)
 exten = do_playback,n,Playback(${CfMC_WhatToPlay})
 ; PLAYBACKSTATUS - SUCCESS FAILED
 exten = do_playback,n,UserEvent(AfterPlayBack,ActionID:${CfMC_ActionID} 
 ${UNIQUEID}  ${CHANNEL}  ${CfMC_WhatToPlay}  ${CfMC_WhoHear} 
 ${PLAYBACKSTATUS})
 exten = do_playback,n,Hangup()

 exten = do_chanspy,1,Answer()
 exten = do_chanspy,n,UserEvent(BeforeChanSpy,ActionID:${CfMC_ActionID} 
 ${UNIQUEID}  ${CHANNEL}  ${CfMC_WhatToPlay}  ${CfMC_WhoHear})
 exten = do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
 exten = do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID} 
 ${UNIQUEID}  ${CHANNEL}  ${CfMC_WhatToPlay}  ${CfMC_WhoHear})
 exten = do_chanspy,n,Hangup()


 --
 Jim Dickenson
 mailto:dicken...@cfmc.com dicken...@cfmc.com

 CfMC
 http://www.cfmc.com/



 On Aug 9, 2010, at 5:19 PM, Gabriel Ortiz Lour wrote:

 Hi all,

   How can I playback a file within an active call?

 I've tried with ChanSpy whisper mode like this (using AMI):

 Action: Originate
 Channel: Local/9...@default
 Priority: 0
 Variable: MSG=test
 Application: ChanSpy
 Data: SIP/1234-123
 Async: 1

 and  in the dialplan:

 [default]
 exten = ,1,Answer()
 exten = ,n,Wait(2)
 exten = ,n,Playback(${MSG})

   Where SIP/1234-123 is the up bridged channel.

 But this is not working (it seams that will work on the rolling CLI, but no
 sound at all)

 Is there a better way to do it?
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Re: [asterisk-users] Playback during call

[Jim Dickenson]

Your ami packet is not setting the w option for chanspy, nor I am sure you can 
do this.

You might want to create an additional exten that takes a variable from your 
ami packet and does the chanspy that way.

I use an ami packet like this with extension that do the work.

Action: Originate
Channel: Local/do_playb...@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=PlayBack
Variable: CfMC_WhatToPlay=lyrics-louie-louie
Variable: CfMC_WhoHear=SIP/GXP280_18-0002
ActionID: PlayBack
Async: true


exten = do_playback,1,Answer()
exten = do_playback,n,UserEvent(BeforePlayBack,ActionID:${CfMC_ActionID}  
${UNIQUEID}  ${CHANNEL}  ${CfMC_WhatToPlay}  ${CfMC_WhoHear})
exten = do_playback,n,Wait(0.3)
exten = do_playback,n,Playback(${CfMC_WhatToPlay})
; PLAYBACKSTATUS - SUCCESS FAILED
exten = do_playback,n,UserEvent(AfterPlayBack,ActionID:${CfMC_ActionID}  
${UNIQUEID}  ${CHANNEL}  ${CfMC_WhatToPlay}  ${CfMC_WhoHear}  
${PLAYBACKSTATUS})
exten = do_playback,n,Hangup()

exten = do_chanspy,1,Answer()
exten = do_chanspy,n,UserEvent(BeforeChanSpy,ActionID:${CfMC_ActionID}  
${UNIQUEID}  ${CHANNEL}  ${CfMC_WhatToPlay}  ${CfMC_WhoHear})
exten = do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten = do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID}  
${UNIQUEID}  ${CHANNEL}  ${CfMC_WhatToPlay}  ${CfMC_WhoHear})
exten = do_chanspy,n,Hangup()


-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Aug 9, 2010, at 5:19 PM, Gabriel Ortiz Lour wrote:

 Hi all,
 
   How can I playback a file within an active call?
 
 I've tried with ChanSpy whisper mode like this (using AMI):
 
 Action: Originate
 Channel: Local/9...@default
 Priority: 0
 Variable: MSG=test
 Application: ChanSpy
 Data: SIP/1234-123
 Async: 1
 
 and  in the dialplan:
 
 [default]
 exten = ,1,Answer()
 exten = ,n,Wait(2)
 exten = ,n,Playback(${MSG})
 
   Where SIP/1234-123 is the up bridged channel.
 
 But this is not working (it seams that will work on the rolling CLI, but no 
 sound at all)
 
 Is there a better way to do it?
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Re: [asterisk-users] Playback all the sound files

[Jim Dickenson]

What I did what pick up the wav version to my mac and then I can play any of 
them I want in quicktime or any other audio player. Easier for me than cooking 
up some asterisk way.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Apr 23, 2010, at 10:11 AM, Jian Gao wrote:

 Hello.
 
 There are so many sound files in /var/lib/asterisk/en.  Is there an easy 
 way to let me play back all of them one by one while I am watching CLI 
 to see the current file name?
 
 Thanks for help.
 -- 
 Jian Gao
 IT Technician
 SJ Geophysics Ltd. http://www.sjgeophysics.com
 jian@sjgeophysics.com mailto:jian@sjgeophysics.com
 Tel: (604)582-1100
 
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Re: [asterisk-users] Playback all the sound files

[Steve Edwards]

On Fri, 23 Apr 2010, Jian Gao wrote:

 There are so many sound files in /var/lib/asterisk/en. Is there an easy 
 way to let me play back all of them one by one while I am watching CLI 
 to see the current file name?

No.

How about:

for F in /var/lib/asterisk/en/*.wav
do
echo $F
play -q $F
read -p 'Next? '
done

-- 
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] Playback all the sound files

[Danny Nicholas]

This is tested in 1.4.30
#!/usr/bin/perl
#
# allfiles 1.0
#
# reads all of the files in /var/lib/asterisk/sounds/en and plays on the
line
# while showing name on console (CLI)
# save as at: /var/lib/asterisk/agi-bin/allfiles.agi  Be sure to chmod +x
it!
#
# Invocation Example:
#   exten = 68742,1,Answer()
#   exten = 68742,2,agi,allfiles.agi|/var/lib/asterisk/sounds/en
#   exten = 68742,3,Hangup()
#
use strict;

$|=1;

# Setup some variables
my %AGI; my $tests = 0; my $fail = 0; my $pass = 0;
my @masterCacheList = ();
my $maxNumber = 10;

# setup options
my $FILEDIR = $ARGV[0];
if (! $FILEDIR ) {
   $FILEDIR = '/var/lib/asterisk/sounds/en';
   }

  opendir (SOUNDDIR, $FILEDIR) or
  die Cannot open report dir $FILEDIR.;
   while (defined(my $file = readdir(SOUNDDIR))) {
  next if $file !~ /\.gsm/;
  print STDERR play file $file\n;
  my @split_file_name = split /\./, $file;
  my $playfile = $split_file_name[0];
  print STDOUT EXEC BACKGROUND $FILEDIR/$playfile\n;
  my $result = STDIN; check_result($result);
  }
   closedir (SOUNDDIR) or
  die Can't close report dir $FILEDIR.;

exit 0;
sub check_result {
   my ($res) = @_;
   my $retval;
   $tests++;
   if ($res =~ /^200/) {
  return 1;
  }
   else {
  return 0;
  }
   }

Don't know how it will post, but PERL doesn't really care about the spacing
and it's just a QD.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jian Gao
Sent: Friday, April 23, 2010 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Playback all the sound files

Hello.

There are so many sound files in /var/lib/asterisk/en.  Is there an easy 
way to let me play back all of them one by one while I am watching CLI 
to see the current file name?

Thanks for help.
-- 
Jian Gao
IT Technician
SJ Geophysics Ltd. http://www.sjgeophysics.com
jian@sjgeophysics.com mailto:jian@sjgeophysics.com
Tel: (604)582-1100

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Re: [asterisk-users] Playback all the sound files

[Jian Gao]

Thanks for all the help.

For now I am going to use Jim's method - get the wav files to my PC.

Since I don't have wav sound files installed in my * box (we use g729), 
so I went to Digium site downloaded the gz sound files. After unzip them 
I found there is a core-sounds-en.txt  and extra-sounds-en.txt that 
have all the sounds print put word by word. This is even better for me 
as a  non-native English speaker. :)

Cheers,

Jim Dickenson wrote:
 What I did what pick up the wav version to my mac and then I can play any of 
 them I want in quicktime or any other audio player. Easier for me than 
 cooking up some asterisk way.
   

-- 
Jian Gao
IT Technician
SJ Geophysics Ltd. http://www.sjgeophysics.com
jian@sjgeophysics.com mailto:jian@sjgeophysics.com
Tel: (604)582-1100

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Re: [asterisk-users] Playback in h extension

[Danny Nicholas]

Not possible.  H exten is called by a hangup.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Friday, March 05, 2010 8:18 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Playback in h extension

 

Hi people, I'm trying to execute the PlayBack command in the h extension...
but it is not played... is it possible to do that?

Thanks,

 

Anahi

  _  

 

Anahi Ludueña

 

 

 

  _  

¿Te gustaría tener Hotmail en tu móvil Movistar? ¡Es gratis!
http://serviciosmoviles.es.msn.com/hotmail/movistar-particulares.aspx 

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Re: [asterisk-users] Playback in h extension

[Christian Victor]

2010/3/5 Danny Nicholas da...@debsinc.com:
 Not possible.  H exten is called by a hangup.

Well - sometimes not both parties hang up at the same time. ;-) If you
want to play something to the originating party after die Dial()ed
party hangs up use the option g in the Dial command to get more
commands executed after the called party hangs up. There you could
check the system variable DIALSTATUS to check if the called party
ANSWERed the call or was BUSY etc.

I hope that helps a bit. I just wrote it from the back of my mind.
Please check the documentation of the Dial command.

If you are not in a Dial() situation Danny's comment applies. ;-)

Chris

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Re: [asterisk-users] Playback to channel using AMI

[Jim Dickenson]

Here is the AMI packet I use to do this:

Action: Originate
Channel: Local/do_playb...@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=callE1330
Variable: CfMC_WhatToPlay=cfmc/song
Variable: CfMC_WhoHear=SIP/GXP280-16-0844e290
ActionID: callE1330
Async: true


And here are the extensions from extensions.conf: (Watch email line wraps)

exten = do_playback,1,Answer()
exten = do_playback,n,UserEvent(BeforePlayBack,ActionID:${CfMC_ActionID} 
${UNIQUEID}  ${CHANNEL}  ${CfMC_WhatToPlay}  ${CfMC_WhoHear})
exten = do_playback,n,Wait(0.3)
exten = do_playback,n,Playback(${CfMC_WhatToPlay})
; PLAYBACKSTATUS - SUCCESS FAILED
exten = do_playback,n,UserEvent(AfterPlayBack,ActionID:${CfMC_ActionID} 
${UNIQUEID}  ${CHANNEL}  ${CfMC_WhatToPlay}  ${CfMC_WhoHear} 
${PLAYBACKSTATUS})
exten = do_playback,n,Hangup()

exten = do_chanspy,1,Answer()
exten = do_chanspy,n,UserEvent(BeforeChanSpy,ActionID:${CfMC_ActionID} 
${UNIQUEID}  ${CHANNEL}  ${CfMC_WhatToPlay}  ${CfMC_WhoHear})
exten = do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten = do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID} 
${UNIQUEID}  ${CHANNEL}  ${CfMC_WhatToPlay}  ${CfMC_WhoHear})
exten = do_chanspy,n,Hangup()



What this does is dial an extension to play what I want to play and then
bridge in a ChanSpy with whisper so the extension I want to hear the sound
can listen in.

This seems to do the trick for me.

-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



 From: Jon Morgan jmor...@c-a-solutions.co.uk
 Organization: Complete Automotive Solutions
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Date: Wed, 13 May 2009 11:46:04 +0100
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 asterisk-users@lists.digium.com
 Subject: [asterisk-users] Playback to channel using AMI
 
 Hi All,
 
 I was wondering if there's any way in Asterisk 1.4.21.2 to playback a wav
 file to a channel using the AMI?
 
 I've had a play and, as there wasn't a Playback command implemented directly
 in the AMI, I thought about maybe calling an AGI script from the AMI to do
 this but it seems there's no support for executing AGI through the AMI
 either?
 
 All I found so far was this:
 
 
 http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/
 
 After trying to use Action: AGI through the AMI I discovered the
 functionality hasn't been included in the release (Invalid Command in the
 response) and wondered why?
 
 Is there any other way to playback a wav to a channel using AMI?
 
 Regards,
 
 Jon Morgan.
 
 
 
 
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Re: [asterisk-users] Playback to channel using AMI

[Jon Morgan]

That's superb, thanks very much Jim.

J.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: 13 May 2009 15:54
To: Asterisk User MailList
Subject: Re: [asterisk-users] Playback to channel using AMI

Here is the AMI packet I use to do this:

Action: Originate
Channel: Local/do_playb...@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=callE1330
Variable: CfMC_WhatToPlay=cfmc/song
Variable: CfMC_WhoHear=SIP/GXP280-16-0844e290
ActionID: callE1330
Async: true


And here are the extensions from extensions.conf: (Watch email line wraps)

exten = do_playback,1,Answer()
exten = do_playback,n,UserEvent(BeforePlayBack,ActionID:${CfMC_ActionID} 
${UNIQUEID}  ${CHANNEL}  ${CfMC_WhatToPlay}  ${CfMC_WhoHear})
exten = do_playback,n,Wait(0.3)
exten = do_playback,n,Playback(${CfMC_WhatToPlay})
; PLAYBACKSTATUS - SUCCESS FAILED
exten = do_playback,n,UserEvent(AfterPlayBack,ActionID:${CfMC_ActionID} 
${UNIQUEID}  ${CHANNEL}  ${CfMC_WhatToPlay}  ${CfMC_WhoHear} 
${PLAYBACKSTATUS})
exten = do_playback,n,Hangup()

exten = do_chanspy,1,Answer()
exten = do_chanspy,n,UserEvent(BeforeChanSpy,ActionID:${CfMC_ActionID} 
${UNIQUEID}  ${CHANNEL}  ${CfMC_WhatToPlay}  ${CfMC_WhoHear})
exten = do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten = do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID} 
${UNIQUEID}  ${CHANNEL}  ${CfMC_WhatToPlay}  ${CfMC_WhoHear})
exten = do_chanspy,n,Hangup()



What this does is dial an extension to play what I want to play and then
bridge in a ChanSpy with whisper so the extension I want to hear the sound
can listen in.

This seems to do the trick for me.

-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



 From: Jon Morgan jmor...@c-a-solutions.co.uk
 Organization: Complete Automotive Solutions
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Date: Wed, 13 May 2009 11:46:04 +0100
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 asterisk-users@lists.digium.com
 Subject: [asterisk-users] Playback to channel using AMI
 
 Hi All,
 
 I was wondering if there's any way in Asterisk 1.4.21.2 to playback a wav
 file to a channel using the AMI?
 
 I've had a play and, as there wasn't a Playback command implemented
directly
 in the AMI, I thought about maybe calling an AGI script from the AMI to do
 this but it seems there's no support for executing AGI through the AMI
 either?
 
 All I found so far was this:
 
 
 http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/
 
 After trying to use Action: AGI through the AMI I discovered the
 functionality hasn't been included in the release (Invalid Command in the
 response) and wondered why?
 
 Is there any other way to playback a wav to a channel using AMI?
 
 Regards,
 
 Jon Morgan.
 
 
 
 
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No virus found in this incoming message.
Checked by AVG - www.avg.com 
Version: 8.5.329 / Virus Database: 270.12.27/2111 - Release Date: 05/13/09
07:04:00


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Re: [asterisk-users] Playback using AMI

[Danny Nicholas]

Just set up a new spy in the dialplan that performs a Background on the
sound file, then hangs up.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Dickenson
Sent: Thursday, November 20, 2008 4:34 PM
To: Asterisk User MailList
Subject: [asterisk-users] Playback using AMI

Is there a way to inject sound from a sound file into an established call
using AMI?

I have an established call from which I can record either or both legs. I
can additionally spy on the call. Is there any way I can play a sound file
into the call and not loose the ability for the people to continue talking
while listening to the sound file?
-- 
Jim Dickenson
mailto:[EMAIL PROTECTED]

CfMC
http://www.cfmc.com/




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Re: [asterisk-users] Playback using AMI

[Jim Dickenson]

I am not sure how one would do that. If I have a dialplan that does a
chanspy the dialplan hangs on that step so how could you do a background
step?

I am clearly missing something in your suggestion.

I am using version 1.6.0.1 if that makes a difference.

Here is more about my setup.

One leg of the call is created by a logged in agent being added to a queue.
While the other leg of the call is the result of an originate action that
queues the call when it is answered.

I use the monitor action to record either or both legs of the call.

I use an originate action to a third leg that executes a dialplan that does
a chanspy to one of the active legs to spy on the call.

What do you suggest I use to play a sound file so that the original two
legs, or all three legs if there is spying going on, can hear it?

The monitor, spy and playing of sound files are all independent things that
I might want to perform with the original two legs.
-- 
Jim Dickenson
mailto:[EMAIL PROTECTED]

CfMC
http://www.cfmc.com/



 From: Danny Nicholas [EMAIL PROTECTED]
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Date: Thu, 20 Nov 2008 16:46:54 -0600
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Playback using AMI
 
 Just set up a new spy in the dialplan that performs a Background on the
 sound file, then hangs up.
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Jim Dickenson
 Sent: Thursday, November 20, 2008 4:34 PM
 To: Asterisk User MailList
 Subject: [asterisk-users] Playback using AMI
 
 Is there a way to inject sound from a sound file into an established call
 using AMI?
 
 I have an established call from which I can record either or both legs. I
 can additionally spy on the call. Is there any way I can play a sound file
 into the call and not loose the ability for the people to continue talking
 while listening to the sound file?
 -- 
 Jim Dickenson
 mailto:[EMAIL PROTECTED]
 
 CfMC
 http://www.cfmc.com/
 
 
 
 
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Re: [asterisk-users] Playback don't play the beginning if a sound file

[Yves Räber]

It seems this has something to do with the Wait() before the Playback
(Background behaves the same). 

If I remove the Wait, the next Playback is just fine, otherwise it
truncates the beginning of the message.

On Mon, 2008-05-05 at 10:41 +0200, Yves Räber wrote:
 Hello,
 
 I'm using this dialplan to let user record messages. The recording part
 works quite fine, but there is something strange :
 
 When Asterisk plays vm-torerecord, it misses the beginning, I only hear
 the few last seconds (vm-torerecord is a sound file that was in the
 asterisk-sounds cvs repo, but I simply renamed it). 
 
 I've looked on voip-info.org, googled anything I could think about and
 checked on bugs.digium.com, I don't have any clue of what's going on.
 
 Does anyone has an idea ? Thanks.
 
 
 Here is my dialplan :
 
 [record]
 exten = s,1,Answer
 exten = s,n,Set(counter=1)
 exten = s,n,NoOp(${counter})
 exten = s,n,GotoIf($[${counter} = 1]?record)
 exten = s,n(next),System(/bin/rm
 -f /var/lib/asterisk/sounds/${RECORDED_FILE}.wav)
 exten = s,n(record),Set(counter=$[${counter}+1]);
 exten = s,n,GotoIf($[${counter}  3]?i,1)
 exten = s,n,Playback(vm-intro)
 exten = s,n,Record(webrecord%d:wav,10,60)
 exten = s,n,Wait(1)
 exten = s,n,Set(CDR(userfield)=${RECORDED_FILE})
 exten = s,n,Playback(${RECORDED_FILE})
 exten = s,n(askretry),Background(vm-torerecord)
 exten = s,n,WaitExten(5)
 exten = i,1,Goto(s,askretry)
 exten = 3,1,Goto(s,next)
 exten = t,1,Set(CDR(userfield)=${RECORDED_FILE})
 exten = t,n,Hangup
 
 
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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Benjamin Jacob]

http://www.openvox.com.cn/products_detail.php?genre_id=9id=28
 
 If you can get the bare card, you can use it for
 timing with a little
 magic that can be found via google.  If not, get one
 with an FXO or
 FXS and you will add a little flexibility and have
 real hardware
 timing.
 
 If you continue to have issues, then you can
 eliminate timing and
 focus on processes I would think.  I had a client
 running spamassassin
 on their Asterisk box which doubled as their
 corporate email server,
 geewhiz, I wonder why they were having issues.
 
 Another odd thing Tzafrir helped me to notice was (I
 don't remember
 what version of CentOS) that the time was jumping
 ahead a couple of
 minutes and then back.  Running top, you could tell
 something was up
 because it was refreshing way too fast.  Then typing
 date on the
 command line repeatedly showed the time jumping all
 over the place.
 Might want to check that out too.
 
 Thanks,
 Steve Totaro
 

Thanks again guys.
the 'watch -d -n 1 cat /proc/interrupts' showed things
to be ok.. the rtc cycles increasing by 1024+ per
second.

In the process of cleaning up unnecesary processes, I
came across this line :

/usr/sbin/vmware-guestd --background
/var/run/vmware-guestd.pid

GASP so does this mean this is a virtual machine??
I have got no idea about virtualization yet. So how do
I confirm if this is a virtual machine or not??

And is it advised to run asterisk on a virtual
machine?

- Ben.



  

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know-it-all with Yahoo! Mobile.  Try it now.  
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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Steve Totaro]

On Mon, Apr 28, 2008 at 3:18 AM, Benjamin Jacob [EMAIL PROTECTED] wrote:



  
  http://www.openvox.com.cn/products_detail.php?genre_id=9id=28
  
   If you can get the bare card, you can use it for
   timing with a little
   magic that can be found via google.  If not, get one
   with an FXO or
   FXS and you will add a little flexibility and have
   real hardware
   timing.
  
   If you continue to have issues, then you can
   eliminate timing and
   focus on processes I would think.  I had a client
   running spamassassin
   on their Asterisk box which doubled as their
   corporate email server,
   geewhiz, I wonder why they were having issues.
  
   Another odd thing Tzafrir helped me to notice was (I
   don't remember
   what version of CentOS) that the time was jumping
   ahead a couple of
   minutes and then back.  Running top, you could tell
   something was up
   because it was refreshing way too fast.  Then typing
   date on the
   command line repeatedly showed the time jumping all
   over the place.
   Might want to check that out too.
  
   Thanks,
   Steve Totaro
  

  Thanks again guys.
  the 'watch -d -n 1 cat /proc/interrupts' showed things
  to be ok.. the rtc cycles increasing by 1024+ per
  second.

  In the process of cleaning up unnecesary processes, I
  came across this line :

  /usr/sbin/vmware-guestd --background
  /var/run/vmware-guestd.pid

  GASP so does this mean this is a virtual machine??
  I have got no idea about virtualization yet. So how do
  I confirm if this is a virtual machine or not??

  And is it advised to run asterisk on a virtual
  machine?



  - Ben.

Ben,

Who is providing your server?  I assume it is in a colo.  Ask them or
see if they mention it in their agreement or sales material.  Finally,
you can just ask them.  If they claimed a dedicated server, complain.

It seems you are running on a virtual machine and no, that is not advisable.

Thanks,
Steve Totaro

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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Benjamin Jacob]

-
   In the process of cleaning up unnecesary
 processes, I
   came across this line :
 
   /usr/sbin/vmware-guestd --background
   /var/run/vmware-guestd.pid
 
   GASP so does this mean this is a virtual
 machine??
   I have got no idea about virtualization yet. So
 how do
   I confirm if this is a virtual machine or not??
 
   And is it advised to run asterisk on a virtual
   machine?
 
 
 
   - Ben.
 
 Ben,
 
 Who is providing your server?  I assume it is in a
 colo.  Ask them or
 see if they mention it in their agreement or sales
 material.  Finally,
 you can just ask them.  If they claimed a dedicated
 server, complain.
 
 It seems you are running on a virtual machine and
 no, that is not advisable.
 
 Thanks,
 Steve Totaro
 


Steve,
Naa.. it's not co-lo. It's a dedicated server for
sure, but my client wants to make the most out of one
box, it seems. 
Talked to the client today and confirmed that it is
indeed a virtual machine. They said they had
previously installed asterisk around a year back on a
virtual machine with no issues. I did not have any
solid convincing response to that. I do understand
about virtualization not being a recommended thing to
do. Now to convince the client.

Also, if I put in a fxo/fxs card, i've read somewhere
that virtual machines won't be able to access the card
n hence the timing provided by it. Is it true?

thanks again Steve and the rest of you.

- Ben.



  

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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Benjamin Jacob]

 
 OK, I think you need to home in on the differences
 between the server(s)
 that work fine and the one that doesn't.

As I said in my other mail, the faulty one is a 
.. mono processor machine, with SMP turned on
.. running CentOS 5
.. with kernel : 2.6.18-53.1.13.el5
There are other kernels too(2.6.18-8, etc.), will be
trying those kernels too.

The local working machine is :
.. dual processor, with SMP ofcourse
.. running Fedora Core 7, if I remember it correctly.
.. kernel definitely  2.6.13

Have looked at all parameters, be it the kernel timer
frequency(1000 HZ), enhanced timer support, etc.
Everything seems to be set right. (Then again, I hope
I am looking at the correct places, i.e. .config files
and using make menuconfig).

 Try watch -dn 1 cat /proc/interrupts and check
 that the RTC interrupts
 are going up by 1024 per second. This is with
 ztdummy running.
This I gotta try. What if it isn't? And worse, what if
it is and I am still getting the choppy playbacks!! 


 
 What else is going on on this server? Does it have
 any virtual machines
 on it? Does it have X Windows running? What does
 top show?

Unfortunately a lot of other processes are running too
on the server, one of them being httpd and other
sundry needed by the client (this inspite of
suggesting him to otherwise).

This is an Asterisk install not done by me, I just
added the zaptel installation and ztdummy module. Was
brazenly confident of things working in a jiffy(does
this count as a pun?), when I stepped in.

cheerz :-(

- Ben.






  

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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Steve Totaro]

On Sat, Apr 26, 2008 at 9:14 AM, Benjamin Jacob [EMAIL PROTECTED] wrote:
 
   OK, I think you need to home in on the differences
   between the server(s)
   that work fine and the one that doesn't.

  As I said in my other mail, the faulty one is a
  .. mono processor machine, with SMP turned on
  .. running CentOS 5
  .. with kernel : 2.6.18-53.1.13.el5
  There are other kernels too(2.6.18-8, etc.), will be
  trying those kernels too.

  The local working machine is :
  .. dual processor, with SMP ofcourse
  .. running Fedora Core 7, if I remember it correctly.
  .. kernel definitely  2.6.13

  Have looked at all parameters, be it the kernel timer
  frequency(1000 HZ), enhanced timer support, etc.
  Everything seems to be set right. (Then again, I hope
  I am looking at the correct places, i.e. .config files
  and using make menuconfig).


   Try watch -dn 1 cat /proc/interrupts and check
   that the RTC interrupts
   are going up by 1024 per second. This is with
   ztdummy running.
  This I gotta try. What if it isn't? And worse, what if
  it is and I am still getting the choppy playbacks!!



  
   What else is going on on this server? Does it have
   any virtual machines
   on it? Does it have X Windows running? What does
   top show?

  Unfortunately a lot of other processes are running too
  on the server, one of them being httpd and other
  sundry needed by the client (this inspite of
  suggesting him to otherwise).

  This is an Asterisk install not done by me, I just
  added the zaptel installation and ztdummy module. Was
  brazenly confident of things working in a jiffy(does
  this count as a pun?), when I stepped in.

  cheerz :-(


  - Ben.

http://www.openvox.com.cn/products_detail.php?genre_id=9id=28

If you can get the bare card, you can use it for timing with a little
magic that can be found via google.  If not, get one with an FXO or
FXS and you will add a little flexibility and have real hardware
timing.

If you continue to have issues, then you can eliminate timing and
focus on processes I would think.  I had a client running spamassassin
on their Asterisk box which doubled as their corporate email server,
geewhiz, I wonder why they were having issues.

Another odd thing Tzafrir helped me to notice was (I don't remember
what version of CentOS) that the time was jumping ahead a couple of
minutes and then back.  Running top, you could tell something was up
because it was refreshing way too fast.  Then typing date on the
command line repeatedly showed the time jumping all over the place.
Might want to check that out too.

Thanks,
Steve Totaro

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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Tony Mountifield]

In article [EMAIL PROTECTED],
Benjamin Jacob [EMAIL PROTECTED] wrote:
 
 One on my clients' machine had Asterisk 1.4.4. installed. The complained of 
 choppy Playback
 of gsm files.
 So scouring the internet gave me the solution of installing ztdummy and 
 loading it as a module.
 Did it (using zaptel-1.4.1) , but to no effect. Re-compiled asterisk and 
 re-installed. Sill
 no effect.
 
 Do I have to specify any parameter in the Asterisk compilation to look at 
 ztdummy/rtc? As
 far as I remember (am coming back to Asterisk after quite some time now), you 
 don't really
 need to set anything over there for any zaptel specific compilation?
 
 And yes, all the files are gsm files and the codec used for the calls is ulaw.
 
 I even tried converting those gsm files to wav using sox and then playing 
 them, but the
 behaviour is the same.
 
 Any ideas anyone.. something I am missing ??

Firstly, check whether Asterisk has chan_zap loaded and access to zaptel:

*CLI zap show channels
   Chan Extension  Context Language   MusicOnHold 
 pseudodefault
*CLI

If you don't get pseudo shown, then you are not getting the benefit of
ztdummy.

However, the probably main cause of choppy sound is poor timing from the
SIP client (I'm assuming SIP), because Asterisk by default uses the incoming
stream to generate timing for the outbound stream.

There are two main things to try:

1. Make sure that the SIP clients are NOT using silence suppression (may
be referred to as VAD, bandwidth saving, or something similar).

2. If ztdummy is running ok, edit /etc/asterisk/asterisk.conf and enable
the line internal_timing=yes. That should make it play out based on
internal zaptel timing instead of timing off the incoming stream, I think.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Doug Lytle]

Tony Mountifield wrote:
 2. If ztdummy is running ok, edit /etc/asterisk/asterisk.conf and enable
 the line internal_timing=yes. That should make it play out based on
   

One other thing comes to mind, make sure you compile with 'Don't 
optimize' if you're using gcc 4.2.2

Doug

-- 
 
Ben Franklin quote:

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Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Benjamin Jacob]

Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED],
Benjamin Jacob  wrote:
 
 One on my clients' machine had Asterisk 1.4.4. installed. The complained of 
 choppy Playback
 of gsm files.
 So scouring the internet gave me the solution of installing ztdummy and 
 loading it as a module.
 Did it (using zaptel-1.4.1) , but to no effect. Re-compiled asterisk and 
 re-installed. Sill
 no effect.
 
 Do I have to specify any parameter in the Asterisk compilation to look at 
 ztdummy/rtc? As
 far as I remember (am coming back to Asterisk after quite some time now), you 
 don't really
 need to set anything over there for any zaptel specific compilation?
 
 And yes, all the files are gsm files and the codec used for the calls is ulaw.
 
 I even tried converting those gsm files to wav using sox and then playing 
 them, but the
 behaviour is the same.
 
 Any ideas anyone.. something I am missing ??

Firstly, check whether Asterisk has chan_zap loaded and access to zaptel:

*CLI zap show channels
   Chan Extension  Context Language   MusicOnHold 
 pseudodefault
*CLI

If you don't get pseudo shown, then you are not getting the benefit of
ztdummy.

However, the probably main cause of choppy sound is poor timing from the
SIP client (I'm assuming SIP), because Asterisk by default uses the incoming
stream to generate timing for the outbound stream.

There are two main things to try:

1. Make sure that the SIP clients are NOT using silence suppression (may
be referred to as VAD, bandwidth saving, or something similar).

2. If ztdummy is running ok, edit /etc/asterisk/asterisk.conf and enable
the line internal_timing=yes. That should make it play out based on
internal zaptel timing instead of timing off the incoming stream, I think.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

Thanks Tony for the response.
zap show channels shows that things are fine, as you said :
*CLI zap show channels
   Chan Extension  Context Language   MOH Interpret   
 pseudodefaultdefault

Tried setting internal_timing to yes as well. Still no difference.

Also,  I don't think my SIP gateway uses Silence suppression, because the same 
SIP gateway connections work fine with another Asterisk server.

This is getting seriously irritating now!!! Have tried all the tricks and tips 
I've been finding on the net.

Yeah, btw, even Meetme playback is choppy. So, I think its somehow related to 
timing. But I am not the expert. 

- Ben.

   
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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Tony Mountifield]

In article [EMAIL PROTECTED],
Benjamin Jacob [EMAIL PROTECTED] wrote:
 
 Also,  I don't think my SIP gateway uses Silence suppression, because the 
 same SIP gateway
 connections work fine with another Asterisk server.

OK, I think you need to home in on the differences between the server(s)
that work fine and the one that doesn't.

What version of kernel is it running? If it less than 2.6.13, make sure
you change #if 0 to #if 1 in ztdummy.c so that USE_RTC still gets
enabled.

Try watch -dn 1 cat /proc/interrupts and check that the RTC interrupts
are going up by 1024 per second. This is with ztdummy running.

What else is going on on this server? Does it have any virtual machines
on it? Does it have X Windows running? What does top show?

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Tzafrir Cohen]

On Fri, Apr 25, 2008 at 03:02:14PM +, Tony Mountifield wrote:
 In article [EMAIL PROTECTED],
 Benjamin Jacob [EMAIL PROTECTED] wrote:
  
  Also,  I don't think my SIP gateway uses Silence suppression, because the 
  same SIP gateway
  connections work fine with another Asterisk server.
 
 OK, I think you need to home in on the differences between the server(s)
 that work fine and the one that doesn't.
 
 What version of kernel is it running? If it less than 2.6.13, make sure
 you change #if 0 to #if 1 in ztdummy.c so that USE_RTC still gets
 enabled.

RTC is available (and used) as of kernel 2.6.15 . The thing that has
changed in 2.6.13 is that the default of HZ became 250 (but still
tunable). So unless you build your own kernel, without using RTC you
would not really get a steady rate of 1000 interrupts per second.

And then again, on kernels = 2.6.22 you have hi-resolution timers which
generally work better.

 
 Try watch -dn 1 cat /proc/interrupts and check that the RTC interrupts
 are going up by 1024 per second. This is with ztdummy running.

And if using something other than RTC: 1000 interrupts per second.
Anyway, close to 1000 is easy to spot there.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Benjamin Jacob]

Benjamin Jacob [EMAIL PROTECTED] wrote: 

Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED],
Benjamin Jacob  wrote:
 
 One on my clients' machine had Asterisk 1.4.4. installed. The complained of 
 choppy Playback
 of gsm files.
 So scouring the internet gave me the solution of installing ztdummy and 
 loading it as a module.
 Did it (using zaptel-1.4.1) , but to no effect. Re-compiled asterisk and 
 re-installed. Sill
 no effect.
 
 Do I have to specify any parameter in the Asterisk compilation to look at 
 ztdummy/rtc? As
 far as I remember (am coming back to Asterisk after quite some time now), you 
 don't really
 need to set anything over there for any zaptel specific compilation?
 
 And yes, all the files are  gsm files and the codec used for the calls is 
 ulaw.
 
 I even tried converting those gsm files to wav using sox and then playing 
 them, but the
 behaviour is the same.
 
 Any ideas anyone.. something I am missing ??

Firstly, check whether Asterisk has chan_zap loaded and access to zaptel:

*CLI zap show channels
   Chan Extension  Context Language   MusicOnHold 
 pseudodefault
*CLI

If you don't get pseudo shown, then you are not getting the benefit of
ztdummy.

However, the probably main cause of choppy sound is poor timing from the
SIP client (I'm assuming SIP), because Asterisk by default uses the incoming
stream to generate timing for the outbound stream.

There are two main things to try:

1. Make sure that the SIP clients are NOT using silence suppression (may
be referred to as VAD, bandwidth saving, or something similar).

2. If  ztdummy is running ok, edit /etc/asterisk/asterisk.conf and enable
the line internal_timing=yes. That should make it play out based on
internal zaptel timing instead of timing off the incoming stream, I think.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

Thanks Tony for the response.
zap show channels shows that things are fine, as you said :
*CLI zap show channels
   Chan Extension  Context Language   MOH Interpret   
 pseudodefaultdefault

Tried setting internal_timing to yes as well. Still  no difference.

Also,  I don't think my SIP gateway uses Silence suppression, because the same 
SIP gateway connections work fine with another Asterisk server.

This is getting seriously irritating now!!! Have tried all the tricks and tips 
I've been finding on the net.

Yeah, btw, even Meetme playback is choppy. So, I think its somehow related to 
timing. But I am not the expert. 

- Ben.
 

-
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   http://lists.digium.com/mailman/listinfo/asterisk-usersBtw, I am on CentOS 
5, with uname showing as:
Linux mserver.org 2.6.18-53.1.13.el5 #1 SMP Tue Feb 12 13:01:45 EST 2008 i686 
i686 i386 GNU/Linux

And it is not a multiprocessor machine. Will the SMP option affect the working 
in any way?

- Ben.

   
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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Tony Mountifield]

In article [EMAIL PROTECTED],
Tzafrir Cohen [EMAIL PROTECTED] wrote:
 
 RTC is available (and used) as of kernel 2.6.15 . The thing that has
 changed in 2.6.13 is that the default of HZ became 250 (but still
 tunable). So unless you build your own kernel, without using RTC you
 would not really get a steady rate of 1000 interrupts per second.

Well, I'm not familiar with the later 2.6 kernels (most of my systems
are at 2.6.12 (FC3) or 2.6.9 (RHEL4 clones)).

However, the USE_RTC code was my creation, so I'm very familiar with the
issues as they were at the time.  See http://bugs.digium.com/view.php?id=4301

The issue was that the original 2.6 version of ztdummy ran off the 1000Hz
kernel jiffy counter. This had a tendency to miss ticks. Having successfully
used the zaprtc module in 2.4, I re-implemented it for 2.6 using the rtc
hooks that the 2.6 kernel provided. It sets the 146818 RTC chip to generate
1024Hz interrupts (it can't do 1000Hz), and then skips 3 every 128, evenly
spaced. This was a huge improvement over the jiffy counter.

Unfortunately, when the patch was applied to CVS, someone screwed up and
missed out ztdummy.h, only doing ztdummy.c. This broke compilation and
caused BKW to throw a fit. The knee-jerk reaction was to slap an #if 0
around the #define USE_RTC, rather than understand the cause of the problem.
Once ztdummy.h was patched correctly, the #if 0 should have been removed,
but it never was, so most people continued to build it with the inferior
jiffy clock.

When kernel 2.6.13 came along, the jiffy clock no longer defaulted to 1000Hz,
so USE_RTC was made the default for those versions. I will never understand
why it was never just enabled for *all* 2.6 kernels at that time, like it
should have been in the first place.

The only dependency it has is that the kernel must have been built with
CONFIG_RTC and not CONFIG_GENRTC.

 And then again, on kernels = 2.6.22 you have hi-resolution timers which
 generally work better.

I have yet to experience these, but it sounds promising.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] Playback file and detect a key press

[Bob Smither]

On Sat, 2007-12-08 at 13:40 -0600, Moises Silva wrote:
 Bob,
 
 GET DATA should do something like that. But, to do exactly that, you
 could try a patch I did to call AGI(agi:async), this is a special way
 of AGI. As you know, you can already call AGI(name-of-script.php), or
 AGI(agi://ipaddress) for the so named FastAGI. I created another
 variant AGI(agi:async) that will allow you to run AGI thru the manager
 interface. That way you could easily do this in a language like php.

Thanks Moises - I will take a look at your patch, it sounds interesting.

I found a developed a workaround using a standard AGI script.  The
message is played back using Stream File with 0 offset and the allowed
escape digits.  When an allowed digit is pressed, the digit and the file
offset is returned by Stream File.  Using the pressed digit you can go
to an appropriate routine to play back an acknowledgment Thank you for
pressing ... and then start the message playback just before the time
that the key was pressed.

The user hears the message, then has feedback that the key was detected,
and then hears the rest of the message.  This actually works pretty
well.

Thanks!
-- 
Bob Smither, PhD   Circuit Concepts, Inc.
=
   You have the right to YOUR life, liberty and property, but not MINE
=
[EMAIL PROTECTED]  http://www.C-C-I.Com  281-331-2744(office)  -4616(fax)


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Re: [asterisk-users] Playback file and detect a key press

[Moises Silva]

Bob,

GET DATA should do something like that. But, to do exactly that, you
could try a patch I did to call AGI(agi:async), this is a special way
of AGI. As you know, you can already call AGI(name-of-script.php), or
AGI(agi://ipaddress) for the so named FastAGI. I created another
variant AGI(agi:async) that will allow you to run AGI thru the manager
interface. That way you could easily do this in a language like php.

agi_handler.php

/* AGI(agi:async) launch a manager event at start with all the init
data you get usually in a normal AGI (agi_request, agi_exten etc etc)
url-encoded. This function is called anytime that event is received */
function handle_AsyncAGI($data){
$agi_data = urldecode($data);
 $my_manager-SendAction('AGI', array('Channel' =
$agi_data['agi_channel'], 'Command' = 'EXEC Playback tt-monkeys',
'CommandID' = 'IDx'));
}

function handle_EndPlayback($data)
{
 // here you catch the end of the AGI EXEC playback command
executed and you know if DTMF was received on that channel
}

function OnDtmfEvent($data)
{
   // here you can see if the DTMF was received in some channel
executing playback
}


With this you get a lot of power, but also means you need to know how
to do async programming. The script of course, will needs to stay
connected to the manager in order to send the AGI commands and catch
the event responses.

You can find my patch back-ported to 1.4 here:

http://www.moythreads.com/asterisk-1.4.15-async-agi.patch

Also, you can follow the bug in
http://bugs.digium.com/view.php?id=11282 to see if it gets included in
Asterisk 1.6 ( not likely since I don't think Digium is accepting
features for 1.6 ) or a future release.

I will update that bug report with better documentation. Please not
that the original description in the bug does not suit anymore since
my initial patch was different.

Regards,

Moisés Silva

On Dec 8, 2007 12:05 AM, Bob Smither [EMAIL PROTECTED] wrote:
 I would like to do the following:

 Play back a file, and during the playback be able to detect a DTMF tone
 that may be pressed.  I do not want to interrupt the playing of the
 file, but when the file finishes I would like to be able to tell if a
 key was pressed and which key it was.

 Anyway to do this?

 In AGI:

 o  Wait for Digit waits for a digit to be pressed, and I don't see how
 to play a file at the same time.
 o  Stream File can detect a digit, but then the file playback is
 interrupted.

 In a call plan:

 o  Playback plays a file but does not detect pressed digits.
 o  Background plays a file, but stops the playback when a key is
 pressed.

 Is there anyway to do what I want to do?

 Thanks!

 --
 Bob Smither [EMAIL PROTECTED]


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Re: [asterisk-users] Playback a video file?

[Paul Hales]

As far as I know, yes.

Someone even published a how-to on making up video IVR's.

PaulH


On Sun, 2007-08-12 at 08:54 -0400, SIP wrote:
 Is it possible to record or playback a video file in Asterisk?
 
 N.
 
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Re: [asterisk-users] Playback 5% Too Fast?

[Tim Panton]

On 13 Mar 2007, at 00:32, David Brazier wrote:


Hi All

I have a problem with IVR scripts which consist mainly of Playback of
audio files, driven from an AGI application.  There are clicks  
every few
seconds or more frequently that is audible on the remote end  
(PSTN), but

not on the Asterisk recording of the call.  If I record the remote end
and compare it to the local recording, it appears to be about 5%-7%  
too
fast - i.e. if I synchronise the starts, the remote end finishes  
sooner.

I can find points in the remote recording where parts of the waveform
have been missed out, leading to jumps in the waveform, which  
correspond
to the audible clicks.  These jumps seem like dropped packets,  
and I'm

deducing that Asterisk is sending data slightly too fast (i.e. more
frequently than 50x160 sample per second) for the remote end, which  
has

to drop data to keep up.

This is a VoIP-only set up - no Zap hardware.  Thinking this was a
timing issue, I have installed Zaptel to get ztdummy, which is loaded
OK, but that hasn't made any difference.  I have tried it with  
different

VoIP providers and observed the same problem.

Behaviour has persisted from 1.2 to 1.4 and now 1.4.1.  CentOS 4.4
(2.6.9 kernel), Dell 1950.

Any ideas how to progress?  Is this a timing issue or am I wide of the
mark?

Thanks for any help

David


It would be interesting to see an ethereal trace of the packets going
to your PRI gateway. Ideally the packet capture would be done by
a separate system, so that the clock of your Dell won't also be the
'reference' clock.

Do you run NTP on that system. If you do, take a look at the
skew over a day or so and see if the Dell is running fast.

It might be worth investing in a low end digium card
just to generate a clock that is independent of your CPU
clock.

Tim Panton

www.mexuar.net
www.westhawk.co.uk/



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Re: [asterisk-users] Playback 5% Too Fast?

[Cosmin Prund]

I've had similar behavior on my own IVR. I moved my sound files to a ram 
disk and all pops and ticks stopped!


David Brazier wrote:

Hi All

I have a problem with IVR scripts which consist mainly of Playback of
audio files, driven from an AGI application.  There are clicks every few
seconds or more frequently that is audible on the remote end (PSTN), but
not on the Asterisk recording of the call.  If I record the remote end
and compare it to the local recording, it appears to be about 5%-7% too
fast - i.e. if I synchronise the starts, the remote end finishes sooner.
I can find points in the remote recording where parts of the waveform
have been missed out, leading to jumps in the waveform, which correspond
to the audible clicks.  These jumps seem like dropped packets, and I'm
deducing that Asterisk is sending data slightly too fast (i.e. more
frequently than 50x160 sample per second) for the remote end, which has
to drop data to keep up.  


This is a VoIP-only set up - no Zap hardware.  Thinking this was a
timing issue, I have installed Zaptel to get ztdummy, which is loaded
OK, but that hasn't made any difference.  I have tried it with different
VoIP providers and observed the same problem.

Behaviour has persisted from 1.2 to 1.4 and now 1.4.1.  CentOS 4.4
(2.6.9 kernel), Dell 1950.

Any ideas how to progress?  Is this a timing issue or am I wide of the
mark?

Thanks for any help

David
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Re: [asterisk-users] Playback uses channel's language, background doesn't

[Joanna Liza Mariazeta]

Hi Cameron,

Why not automatically set the language that should be use at the beginning.
Set(LANGUAGE()=nz)

Hope that helps.

Best Regards,
Joanna

On 2/28/07, Moises Silva [EMAIL PROTECTED] wrote:


http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetLanguage

There you can found how you can get the current language ( the same
used by playback ), so you can set a local variable to the current
language and use it instead of the blank value

Regards

On 2/26/07, kjcsb [EMAIL PROTECTED] wrote:




 it may be a bug, try creating a simple test script with only 2
 extensions, one with playback the other one with background and see
 how it works, also post here the asterisk version you are using.
 Asterisk 1.2.13

 exten = 98765,1,Playback(to-listen-to-it)
 exten =
 98764,1,Background(to-listen-to-it|m||macro-systemrecording)
 exten = 98763,1,Background(to-listen-to-it)

 -- Executing Playback(SIP/112233-09289b40, to-listen-to-it) in
new
 stack
 -- Playing 'to-listen-to-it' (language 'nz')
 -- Executing Hangup(SIP/112233-09289b40, ) in new stack
   == Spawn extension (116-2000, h, 1) exited non-zero on
 'SIP/112233-09289b40'
 -- Executing BackGround(SIP/112233-09289b40,
 to-listen-to-it|m||macro-systemrecording) in new stack
 -- Playing 'to-listen-to-it' (language '')
 -- Executing Hangup(SIP/112233-09289b40, ) in new stack
   == Spawn extension (116-2000, h, 1) exited non-zero on
 'SIP/112233-09289b40'
 -- Executing BackGround(SIP/112233-09289b40,
 to-listen-to-it) in new stack
 -- Playing 'to-listen-to-it' (language 'nz')
 -- Executing Hangup(SIP/112233-09289b40, ) in new stack
   == Spawn extension (116-2000, h, 1) exited non-zero on
 'SIP/112233-09289b40'

 So it seems assume that since I passed a blank language override to the
 Background application, that I want a blank language. Any ideas on how
to
 get background to use the default language?

 Regards

 Cameron
  
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Re: [asterisk-users] Playback uses channel's language, background doesn't

[kjcsb]

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetLanguage

There you can found how you can get the current language ( the same
used by playback ), so you can set a local variable to the current
language and use it instead of the blank value

This works:
exten = 
98764,1,Background(to-listen-to-it|m|${LANGUAGE()}|macro-systemrecording)

Wiki updated.

Thanks

Cameron



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Re: [asterisk-users] Playback uses channel's language, background doesn't

[Moises Silva]

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetLanguage

There you can found how you can get the current language ( the same
used by playback ), so you can set a local variable to the current
language and use it instead of the blank value

Regards

On 2/26/07, kjcsb [EMAIL PROTECTED] wrote:





it may be a bug, try creating a simple test script with only 2
extensions, one with playback the other one with background and see
how it works, also post here the asterisk version you are using.
Asterisk 1.2.13

exten = 98765,1,Playback(to-listen-to-it)
exten =
98764,1,Background(to-listen-to-it|m||macro-systemrecording)
exten = 98763,1,Background(to-listen-to-it)

-- Executing Playback(SIP/112233-09289b40, to-listen-to-it) in new
stack
-- Playing 'to-listen-to-it' (language 'nz')
-- Executing Hangup(SIP/112233-09289b40, ) in new stack
  == Spawn extension (116-2000, h, 1) exited non-zero on
'SIP/112233-09289b40'
-- Executing BackGround(SIP/112233-09289b40,
to-listen-to-it|m||macro-systemrecording) in new stack
-- Playing 'to-listen-to-it' (language '')
-- Executing Hangup(SIP/112233-09289b40, ) in new stack
  == Spawn extension (116-2000, h, 1) exited non-zero on
'SIP/112233-09289b40'
-- Executing BackGround(SIP/112233-09289b40,
to-listen-to-it) in new stack
-- Playing 'to-listen-to-it' (language 'nz')
-- Executing Hangup(SIP/112233-09289b40, ) in new stack
  == Spawn extension (116-2000, h, 1) exited non-zero on
'SIP/112233-09289b40'

So it seems assume that since I passed a blank language override to the
Background application, that I want a blank language. Any ideas on how to
get background to use the default language?

Regards

Cameron
 
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Re: [asterisk-users] Playback uses channel's language, background doesn't

[Moises Silva]

it may be a bug, try creating a simple test script with only 2
extensions, one with playback the other one with background and see
how it works, also post here the asterisk version you are using.

Regards

On 2/26/07, kjcsb [EMAIL PROTECTED] wrote:


I have the following in the dialplan:
[macro-systemrecording]
exten = s,1,Goto(${ARG1},1)
exten =
dorecord,1,Record(/tmp/${CALLERID(number)}-ivrrecording:wav)
exten = dorecord,n,Wait(1)
exten = dorecord,n,Goto(confmenu,1)
exten =
docheck,1,Playback(/tmp/${CALLERID(number)}-ivrrecording)
exten = docheck,n,Wait(1)
exten = docheck,n,Goto(confmenu,1)
exten =
confmenu,1,Background(to-listen-to-itpress-1to-rerecord-itpress-star|m||macro-systemrecording)
exten = confmenu,n,Read(RECRESULT||1|||4)
exten =
confmenu,n,GotoIf($[x${RECRESULT}=x*]?dorecord,1)
exten =
confmenu,n,GotoIf($[x${RECRESULT}=x1]?docheck,1)
exten = confmenu,n,Goto(1)
exten = 1,1,Goto(docheck,1)
exten = *,1,Goto(dorecord,1)
exten = t,1,Playback(goodbye)
exten = t,n,Hangup
exten = i,1,Playback(pm-invalid-option)
exten = i,n,Goto(confmenu,1)
exten = h,1,Hangup

When this is called the following is shown in the CLI
-- Goto (macro-systemrecording,docheck,1)
-- Executing Playback(SIP/223344-0928bbb8, /tmp/2595-ivrrecording)
in new stack
-- Playing '/tmp/2595-ivrrecording' (language 'nz')
-- Executing Wait(SIP/223344-0928bbb8, 1) in new stack
-- Executing Goto(SIP/223344-0928bbb8, confmenu|1) in new stack
-- Goto (macro-systemrecording,confmenu,1)
-- Executing BackGround(SIP/223344-0928bbb8,
to-listen-to-itpress-1to-rerecord-itpress-star|m||macro-systemrecording)
in new stack
-- Playing 'to-listen-to-it' (language '')

As can be seen, Playback uses the channel's language 'nz' but BackGround
does not. Could anyone advise what I'm doing wrong?

Thanks

Cameron
 
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Re: [asterisk-users] Playback uses channel's language, background doesn't

[kjcsb]

it may be a bug, try creating a simple test script with only 2
extensions, one with playback the other one with background and see
how it works, also post here the asterisk version you are using.
Asterisk 1.2.13 

exten = 98765,1,Playback(to-listen-to-it)
exten = 98764,1,Background(to-listen-to-it|m||macro-systemrecording)
exten = 98763,1,Background(to-listen-to-it)

-- Executing Playback(SIP/112233-09289b40, to-listen-to-it) in new stack
-- Playing 'to-listen-to-it' (language 'nz')
-- Executing Hangup(SIP/112233-09289b40, ) in new stack
  == Spawn extension (116-2000, h, 1) exited non-zero on 
'SIP/112233-09289b40'
-- Executing BackGround(SIP/112233-09289b40, 
to-listen-to-it|m||macro-systemrecording) in new stack
-- Playing 'to-listen-to-it' (language '')
-- Executing Hangup(SIP/112233-09289b40, ) in new stack
  == Spawn extension (116-2000, h, 1) exited non-zero on 
'SIP/112233-09289b40'
-- Executing BackGround(SIP/112233-09289b40, to-listen-to-it) in new 
stack
-- Playing 'to-listen-to-it' (language 'nz')
-- Executing Hangup(SIP/112233-09289b40, ) in new stack
  == Spawn extension (116-2000, h, 1) exited non-zero on 
'SIP/112233-09289b40'

So it seems assume that since I passed a blank language override to the 
Background application, that I want a blank language. Any ideas on how to get 
background to use the default language?

Regards

Cameron



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Re: [Asterisk-Users] Playback welcome message while phones ring, please help

[Gareth Blades]

I believe if you use the new native music on hold feature it always
plays the music on hold starting from the beginning.

On Tue, 2006-06-06 at 11:15, Tommaso Calosi wrote:
 I want that incoming callers to hear a welcome message while the phones 
 ring. I know I can use Dial with the m(class) option to make the same 
 with musiconhold, but the problem is that musiconhold does not start 
 from the beginning of my mp3 file.  If I use Playback or Background, the 
 phones do not ring unless the mp3 file is over...
 
 Any suggestion?
 
 
 Thanks
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RE: [Asterisk-Users] Playback welcome message while phones ring, please help

[Tim Sharp]

Yes it does, I just set our system up that way.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Gareth
Blades
Sent: Tuesday, June 06, 2006 6:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Playback welcome message while phones
ring,please help


I believe if you use the new native music on hold feature it always
plays the music on hold starting from the beginning.

On Tue, 2006-06-06 at 11:15, Tommaso Calosi wrote:
 I want that incoming callers to hear a welcome message while the phones 
 ring. I know I can use Dial with the m(class) option to make the same 
 with musiconhold, but the problem is that musiconhold does not start 
 from the beginning of my mp3 file.  If I use Playback or Background, the 
 phones do not ring unless the mp3 file is over...
 
 Any suggestion?
 
 
 Thanks
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Re: [Asterisk-Users] Playback welcome message while phones ring, please help

[Olivier]

2006/6/6, Gareth Blades [EMAIL PROTECTED]:
I believe if you use the new native music on hold feature it alwaysplays the music on hold starting from the beginning.Where can I find this new native music on hold feature ?
In Asterisk 1.2.x ?Regards
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RE: [Asterisk-Users] Playback welcome message while phones ring, please help

[Tim Sharp]

I am 
running on 1.2.7.1

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of 
  OlivierSent: Tuesday, June 06, 2006 12:17 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] Playback welcome message while phones ring,please 
  help
  2006/6/6, Gareth Blades [EMAIL PROTECTED]:
  I 
believe if you use the new native music on hold feature it alwaysplays 
the music on hold starting from the beginning.Where 
  can I find this "new native music on hold feature" ?In Asterisk 1.2.x 
  ?Regards
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Re: [Asterisk-Users] Playback welcome message while phones ring, please help

[Tommaso Calosi]

Thanks, it works for me too.

Tim Sharp wrote:

Yes it does, I just set our system up that way.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Gareth
Blades
Sent: Tuesday, June 06, 2006 6:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Playback welcome message while phones
ring,please help


I believe if you use the new native music on hold feature it always
plays the music on hold starting from the beginning.

On Tue, 2006-06-06 at 11:15, Tommaso Calosi wrote:
  
I want that incoming callers to hear a welcome message while the phones 
ring. I know I can use Dial with the m(class) option to make the same 
with musiconhold, but the problem is that musiconhold does not start 
from the beginning of my mp3 file.  If I use Playback or Background, the 
phones do not ring unless the mp3 file is over...


Any suggestion?


Thanks
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Re: [Asterisk-Users] playback windows recorded sound

[Akpome Akpoguma]

I downloaded recordPad and recorded a wav file and tried playback on 
asterisk got the same error as before -- WARNING [1225991360] 
Format.wav.c:132 check_header:unexpected header size 18--


when I recorded in gsm format on my laptop asterisk did playback well

I used sox to resample the recorded wav file on the asterisk machine into 
wav again and asterisk playback worked well.


The sound property of the recorded wav file is as follows

Bit Rate128kbps
Audio sample size   16 bit
Channels   1(mono)
Audio sample rate   8 kHz
Audio Format PCM

Is there any reason for this behaviour?



From: Doug Lytle [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com

Subject: Re: [Asterisk-Users] playback windows recorded sound
Date: Mon, 22 May 2006 14:12:25 -0400

Akpome Akpoguma wrote:

Hi guys,

I recorded a wav file on my windows xp laptop and tried to playback on 
asterisk but got the following error..unexpected header error 
18.when  I recorded sound using a sip phone on asterisk and compared 
with what I recorded on windows the sound property looked the 
same.does anyone have an idea how I can resolve this?



This should help:

http://www.voip-info.org/wiki/view/Asterisk+sound+files

Doug

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Re: [Asterisk-Users] playback windows recorded sound

[Time Bandit]

I downloaded recordPad and recorded a wav file and tried playback on
asterisk got the same error as before -- WARNING [1225991360]
Format.wav.c:132 check_header:unexpected header size 18--

when I recorded in gsm format on my laptop asterisk did playback well

I used sox to resample the recorded wav file on the asterisk machine into
wav again and asterisk playback worked well.

The sound property of the recorded wav file is as follows

Bit Rate128kbps
Audio sample size   16 bit
Channels   1(mono)
Audio sample rate   8 kHz
Audio Format PCM

Is there any reason for this behaviour?

See this page :
http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk

hth
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Re: [Asterisk-Users] playback windows recorded sound

[Doug Lytle]

Akpome Akpoguma wrote:

Hi guys,

I recorded a wav file on my windows xp laptop and tried to playback on 
asterisk but got the following error..unexpected header error 
18.when  I recorded sound using a sip phone on asterisk and 
compared with what I recorded on windows the sound property looked the 
same.does anyone have an idea how I can resolve this?



This should help:

http://www.voip-info.org/wiki/view/Asterisk+sound+files

Doug

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Re: [Asterisk-Users] Playback(something,noanswer) on Zap?

[Dmitry Ivanov]

On Thursday 20 April 2006 19:22, Kevin P. Fleming wrote:
 A better solution is to set the PRI hangup cause before dropping the
 incoming call; if you set the hangup cause to 'number not assigned'
 then your telco's switch will play its normal intercept message to
 the caller.

Thank you! This works!

context from-e1 {
_X. = {
AGI(pub2ext.agi);
PRI_CAUSE=1;
Hangup();
};
};

Now caller hears voice from his/her telco (not from my telco) saying 
that number is not available. This is even better.
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Re: [Asterisk-Users] Playback(something,noanswer) on Zap?

[FaberK]

Hi Dmitry,may I ask you if is possible to see the pub2ext.agi code?I'm looking for a solution like your, with no luck since long time(you can see from ml-archive).Thanks a lot!
2006/4/24, Dmitry Ivanov [EMAIL PROTECTED]:
On Thursday 20 April 2006 19:22, Kevin P. Fleming wrote: A better solution is to set the PRI hangup cause before dropping the incoming call; if you set the hangup cause to 'number not assigned' then your telco's switch will play its normal intercept message to
 the caller.Thank you! This works!context from-e1 {_X. = {AGI(pub2ext.agi);PRI_CAUSE=1;Hangup();};};Now caller hears voice from his/her telco (not from my telco) saying
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Re: [Asterisk-Users] Playback(something,noanswer) on Zap?

[Kevin P. Fleming]

Dmitry Ivanov wrote:

 Apparently, Playback(invalid,noanswer) does not work with Zap/PRI. Is 
 this bug?

Yes it does work. However, if your telco will not allow you to send
'early audio', then you can't do it.

A better solution is to set the PRI hangup cause before dropping the
incoming call; if you set the hangup cause to 'number not assigned' then
 your telco's switch will play its normal intercept message to the caller.
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Re: [Asterisk-Users] playback soundfile in memory

[Matt Roth]

Akpome Akpoguma wrote:



I want to playback sound file loaded in memory not from a 
file...is this possible?



Akpome,

If the sound file is being played more than once, there is a good chance 
that this is already happening.  At one point, our production system had 
100 calls in queue.  Each call had pre-queue announcements from the 
dialplan, native MOH, and in-queue announcements.  I ran iostat on the 
system, and there was no disk read activity at all.  I believe this can 
be accounted for by Linux's file caching.


If you run iostat on your system and see read activity, give this a try:

1) Set up a RAM disk
2) Build an init-script that copies your sound files from the hard disk 
to the RAM disk on boot

3) Configure Asterisk to play the files from the RAM disk

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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RE: [Asterisk-Users] playback soundfile stored in mysql database

[Alexander Lopez]

Look at using EAGI.
 
 
 Hi Guys,
 
 I want to playback a sound file stored in mysql database in 
 my perl scriptpls can anyone help with an idea? 
 response would be greatly appreciated
 
 Rgds
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RE: [Asterisk-Users] playback soundfile stored in mysql database

[Akpome Akpoguma]

.want to playback a raw binary file without writing into an 
intermediate file which would increase latency




From: Alexander Lopez [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com

Subject: RE: [Asterisk-Users] playback soundfile stored in mysql database
Date: Wed, 12 Apr 2006 13:17:13 -0400

Look at using EAGI.


 Hi Guys,

 I want to playback a sound file stored in mysql database in
 my perl scriptpls can anyone help with an idea?
 response would be greatly appreciated

 Rgds
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Re: [Asterisk-Users] Playback after Page()

[C F]

I'm not sure it it's going to help for you, but try playing around
with the Local channels. and use that local channel as one of the
called devices in the page app.

On 12/30/05, Douglas Garstang [EMAIL PROTECTED] wrote:
 Reposting. I forgot to change the subject. Oops.


 Just a curiosity really. Anyone know how I can do this?

 exten = page,1,SetVar(_ALERT_INFO=ring-answer)
 exten = page,2,Page(SIP/a00090101SIP/a00090301)
 exten = page,3,Playback(tt-weasels)

 ie Play back the sound file after the phones receiving the page have 
 answered? I know page is really simulating a one-way audio meetme conference, 
 so I don't even know it it's possible.

 Thanks.

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RE: [Asterisk-Users] Playback after Page()

[Alexander Lopez]

You can do something like this:

exten = pagenplay,1,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED])


[page]
exten = _X.,1,Set(TIMEOUT(absolute)=180)   ; Three Minutes
exten = _X.,2,SetVar(_ALERT_INFO=ring-answer)
exten = _X.,3,Dial(SIP/${EXTEN}||A(tt-weasels))


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of C F
 Sent: Friday, December 30, 2005 1:33 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Playback after Page()
 
 I'm not sure it it's going to help for you, but try playing 
 around with the Local channels. and use that local channel as 
 one of the called devices in the page app.
 
 On 12/30/05, Douglas Garstang [EMAIL PROTECTED] wrote:
  Reposting. I forgot to change the subject. Oops.
 
 
  Just a curiosity really. Anyone know how I can do this?
 
  exten = page,1,SetVar(_ALERT_INFO=ring-answer)
  exten = page,2,Page(SIP/a00090101SIP/a00090301)
  exten = page,3,Playback(tt-weasels)
 
  ie Play back the sound file after the phones receiving the 
 page have answered? I know page is really simulating a 
 one-way audio meetme conference, so I don't even know it it's 
 possible.
 
  Thanks.
 
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Re: [Asterisk-Users] Playback before Answer

[Trevor Peirce]

Panitaxx wrote:


I have an ISDN PRI E1. I want to send an audio before answering, I am
using noanswer option in playback app but all the audio is muted
before the answer. I would like to play this audio.
 

I have a T1 and a few months ago my ability to playback audio before 
answering ceased.  Now I just get silence as well.  I imagine a code 
change is responsible but I haven't had the time to go back and figure 
out exactly when this happened.

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Re: [Asterisk-Users] Playback before Answer

[Panitaxx]

It was after an upgrade? This E1 is new. We have  10 E1 R2. This is
our first pri. I am starting to suspect that audio suppresion is done
at the telco.

ia

On 8/11/05, Trevor Peirce [EMAIL PROTECTED] wrote:
 Panitaxx wrote:
 
 I have an ISDN PRI E1. I want to send an audio before answering, I am
 using noanswer option in playback app but all the audio is muted
 before the answer. I would like to play this audio.
 
 
 I have a T1 and a few months ago my ability to playback audio before
 answering ceased.  Now I just get silence as well.  I imagine a code
 change is responsible but I haven't had the time to go back and figure
 out exactly when this happened.
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Re: [Asterisk-Users] Playback() stops working.

[Simon Morris]

On 5/1/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
 
  I'm working on configuring asterisk 1.0.7 on Debian Sarge.
 
  The servers been tested a bit and seemed to working fine, but for some
  reason now, when I try and run the Playback() or Background()
  applications, or even try and goto voicemail asterisk refuses to play
  any sounds back to me.
 
 I have heard from 5 or so people about this problem.  I run CVS STABLE
 (almost the same as 1.0.7) and had none of these issues.  I wish I
 could help you, but I wanted to let you know that this is not a
 general problem.
 

Well a quick reinstall of Asterisk solved the problem, but I hope it
doesn't happen again :)

Thanks for the heads up Eric.

~sm
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Re: [Asterisk-Users] Playback() stops working.

[Robert Derr]

Simon Morris wrote:

  On 5/1/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
  
  

  I'm working on configuring asterisk 1.0.7 on Debian Sarge.

The servers been tested a bit and seemed to working fine, but for some
reason now, when I try and run the Playback() or Background()
applications, or even try and goto voicemail asterisk refuses to play
any sounds back to me.
  

I have heard from 5 or so people about this problem.  I run CVS STABLE
(almost the same as 1.0.7) and had none of these issues.  I wish I
could help you, but I wanted to let you know that this is not a
general problem.


  
  
Well a quick reinstall of Asterisk solved the problem, but I hope it
doesn't happen again :)

Thanks for the heads up Eric.
  

I'm having the same problem. What exactly did you do yo reinstall it?
Just a make  make install?

Thanks
-Rob



begin:vcard
fn:Robert Derr
n:Derr;Robert
org:WeatherFlow, Inc.;IT Florida office
adr:;;120 Canal St;New Smyrna Beach;FL;32168;USA
email;internet:[EMAIL PROTECTED]
title:Software Developer
tel;work:386-423-1516
tel;fax:386-409-5178
url:http://www.iwindsurf.com/
version:2.1
end:vcard

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Re: [Asterisk-Users] Playback() stops working.

[Eric Wieling aka ManxPower]

I'm working on configuring asterisk 1.0.7 on Debian Sarge.
The servers been tested a bit and seemed to working fine, but for some
reason now, when I try and run the Playback() or Background()
applications, or even try and goto voicemail asterisk refuses to play
any sounds back to me.
I have heard from 5 or so people about this problem.  I run CVS STABLE 
(almost the same as 1.0.7) and had none of these issues.  I wish I 
could help you, but I wanted to let you know that this is not a 
general problem.

--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] Playback dosen't play Playtone(Congestion) does ?

[Michael D Schelin]

Tim, I found something in the huge log file.
Apr 25 23:25:51 DEBUG[2330]: Ooh, format changed from unknown to ulaw
what does this mean Ooh ?
to 208.41.254.119:5060
Apr 25 23:25:51 VERBOSE[2330]: -- Executing
Playback(SIP/208.41.254.119-089b23d8, 

telephone-in-your-pocket) in new stack
Apr 25 23:25:51 DEBUG[2330]: Ooh, format changed from unknown to ulaw
Apr 25 23:25:51 DEBUG[2330]: Scheduling timer at 160 sample intervals
Apr 25 23:25:51 VERBOSE[2330]: -- Playing 'telephone-in-your-pocket'
(language 'en')
Apr 25 23:25:51 VERBOSE[2330]:
 I see all log files.  Sip debug is on so I see it wanting to play the 
file.  I'm the root. I was thinking path but but I tested it by changing 
the name to something wrong. The call went right to and error tone. 
When It's correct it's just silence.  The tones are not in the same 
directory as the sounds. Asterisk knows the directory.  I will look at 
the log file.  I have looked at the asterisk.conf  docs. and I can't 
find the Variable need the to put in for the sound files path. Example 
below is my asterisk.conf file. I would like to force it to the correct 
directory.

 [directories]
astetcdir = /etc/asterisk
astmoddir = /usr/lib/asterisk/modules
astvarlibdir = /var/lib/asterisk
astagidir = /var/lib/asterisk/agi-bin
astspooldir = /var/spool/asterisk
astrundir = /var/run/asterisk
astlogdir = /var/log/asterisk




Tim Connolly wrote:
Are you seeing anything in your /var/log/asterisk/messages file or even on
the console with verbosity at 3 or more? I'm guessing you have a path or
permissions problem, but you should see either in the logs or the console.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael D
Schelin
Sent: Monday, April 25, 2005 8:29 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Playback dosen't play Playtone(Congestion) does ?
Hi All, What would keep Asterisk from playing out audio files (Playback 
command) but I can play the busy tone . playtone(Congestion)  ??  I have 
verified this with ethereal and see the audio only going one way.  
Because I can hear the audio with the play tone I know there is 
something preventing the playback from working.

Thanks
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RE: [Asterisk-Users] Playback dosen't play Playtone(Congestion) does ?

[Tim Connolly]

Are you seeing anything in your /var/log/asterisk/messages file or even on
the console with verbosity at 3 or more? I'm guessing you have a path or
permissions problem, but you should see either in the logs or the console.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael D
Schelin
Sent: Monday, April 25, 2005 8:29 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Playback dosen't play Playtone(Congestion) does ?

Hi All, What would keep Asterisk from playing out audio files (Playback 
command) but I can play the busy tone . playtone(Congestion)  ??  I have 
verified this with ethereal and see the audio only going one way.  
Because I can hear the audio with the play tone I know there is 
something preventing the playback from working.

Thanks
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Re: [Asterisk-Users] Playback of sound files but no sound

[Gareth Blades]

The most common cause for this is there being no timing source
available. Do you have the zaptel drivers correctly installed and
configured?
You could just enable 'ztdummy' and test the system using that.

On Wed, 2005-03-23 at 12:02, Bart Van Daal wrote:
 Hello,
 
 I'm running asterisk-1.0.6 on a centos3.4 box. 
 I'm still in testing phase and so far everything is running smoothly.
 I'm now trying to play a soundfile or an mp3file using 'MP3Player',
 'Playback'
 or the 'Background' commands, but don't get any sound.
 The logfile says:
 -- Executing BackGround(SIP/joa-9def, tt-weasels) in new stack
 -- Playing 'tt-weasels' (language 'en')
 Are the sound drivers (alsa or oss) used for this or do I need to configure
 something else?
 
 thanks for any help,
 Bart
 
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RE: [Asterisk-Users] Playback of sound files but no sound

[Bart Van Daal]

Thanks for your answer,

I've compiled and loaded 'ztdummy' but still no sound.
here's the relevant portion of lsmod:
ztdummy 2464   0  (unused)
wcusb  19552   0  (unused)
zaptel178752   0  [ztdummy wcusb]
i810_audio 28824   0  (autoclean)
ac97_codec 16840   0  (autoclean) [i810_audio]
soundcore   6436   2  (autoclean) [i810_audio]
usb-uhci   25740   0  [ztdummy] 
Maybe a irrelevant question but do I need alsa or oss (alsa.conf oss.conf)
to play back these sound files?

thanks again,
Bart



-Original Message-
From: Gareth Blades [mailto:[EMAIL PROTECTED] 
Sent: woensdag 23 maart 2005 13:11
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Playback of sound files but no sound

The most common cause for this is there being no timing source available. Do
you have the zaptel drivers correctly installed and configured?
You could just enable 'ztdummy' and test the system using that.

On Wed, 2005-03-23 at 12:02, Bart Van Daal wrote:
 Hello,
 
 I'm running asterisk-1.0.6 on a centos3.4 box. 
 I'm still in testing phase and so far everything is running smoothly.
 I'm now trying to play a soundfile or an mp3file using 'MP3Player', 
 'Playback'
 or the 'Background' commands, but don't get any sound.
 The logfile says:
 -- Executing BackGround(SIP/joa-9def, tt-weasels) in new stack
 -- Playing 'tt-weasels' (language 'en') Are the sound drivers 
 (alsa or oss) used for this or do I need to configure something else?
 
 thanks for any help,
 Bart
 
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Re: [Asterisk-Users] Playback of sound files but no sound

[Eric Wieling aka ManxPower]

Bart Van Daal wrote:
Thanks for your answer,
I've compiled and loaded 'ztdummy' but still no sound.
here's the relevant portion of lsmod:
ztdummy 2464   0  (unused)
wcusb  19552   0  (unused)
zaptel178752   0  [ztdummy wcusb]
i810_audio 28824   0  (autoclean)
ac97_codec 16840   0  (autoclean) [i810_audio]
soundcore   6436   2  (autoclean) [i810_audio]
usb-uhci   25740   0  [ztdummy] 
Maybe a irrelevant question but do I need alsa or oss (alsa.conf oss.conf)
to play back these sound files?
That's because you do NOT need a Zaptel timer for Playback, Background, 
or MoH.
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RE: [Asterisk-Users] Playback of sound files but no sound

[Bart Van Daal]

That's because someone suggested it earlier on the list. So
I installed the ztdummy driver.
Could you please tell me what is needed to playback sound files?

thanks,
Bart 

-Original Message-
From: Eric Wieling aka ManxPower [mailto:[EMAIL PROTECTED] 
Sent: woensdag 23 maart 2005 14:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Playback of sound files but no sound

Bart Van Daal wrote:
 Thanks for your answer,
 
 I've compiled and loaded 'ztdummy' but still no sound.
 here's the relevant portion of lsmod:
 ztdummy 2464   0  (unused)
 wcusb  19552   0  (unused)
 zaptel178752   0  [ztdummy wcusb]
 i810_audio 28824   0  (autoclean)
 ac97_codec 16840   0  (autoclean) [i810_audio]
 soundcore   6436   2  (autoclean) [i810_audio]
 usb-uhci   25740   0  [ztdummy] 
 Maybe a irrelevant question but do I need alsa or oss (alsa.conf 
 oss.conf) to play back these sound files?

That's because you do NOT need a Zaptel timer for Playback, Background, or
MoH.
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Re: [Asterisk-Users] Playback of sound files but no sound

[Eric Wieling aka ManxPower]

Bart Van Daal wrote:
That's because someone suggested it earlier on the list. So
I installed the ztdummy driver.
Could you please tell me what is needed to playback sound files?
Nothing.  It Just Works.
You call into your Asterisk server, dial the extension for the Playback 
or Background, or whatever and hear the file being played.  Simple. 
Something ELSE is going on here.
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RE: [Asterisk-Users] Playback of sound files but no sound

[Gareth Blades]

Do you have these files :-

[EMAIL PROTECTED] default]# ls -l /dev/zap
total 0
crws-T  1 root root 196, 254 Mar 22 09:21 channel
crws-T  1 root root 196,   0 Mar 22 09:21 ctl
crws-T  1 root root 196, 255 Mar 22 09:21 pseudo
crws-T  1 root root 196, 253 Mar 22 09:21 timer

If you are using udev you need to add some configuration manually so
these files get created. There is a note displayed when making the
zaptel package but it is easy to miss.

On Wed, 2005-03-23 at 13:40, Bart Van Daal wrote:
 Thanks for your answer,
 
 I've compiled and loaded 'ztdummy' but still no sound.
 here's the relevant portion of lsmod:
 ztdummy 2464   0  (unused)
 wcusb  19552   0  (unused)
 zaptel178752   0  [ztdummy wcusb]
 i810_audio 28824   0  (autoclean)
 ac97_codec 16840   0  (autoclean) [i810_audio]
 soundcore   6436   2  (autoclean) [i810_audio]
 usb-uhci   25740   0  [ztdummy] 
 Maybe a irrelevant question but do I need alsa or oss (alsa.conf oss.conf)
 to play back these sound files?
 
 thanks again,
 Bart
 
 
 
 -Original Message-
 From: Gareth Blades [mailto:[EMAIL PROTECTED] 
 Sent: woensdag 23 maart 2005 13:11
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Playback of sound files but no sound
 
 The most common cause for this is there being no timing source available. Do
 you have the zaptel drivers correctly installed and configured?
 You could just enable 'ztdummy' and test the system using that.
 
 On Wed, 2005-03-23 at 12:02, Bart Van Daal wrote:
  Hello,
  
  I'm running asterisk-1.0.6 on a centos3.4 box. 
  I'm still in testing phase and so far everything is running smoothly.
  I'm now trying to play a soundfile or an mp3file using 'MP3Player', 
  'Playback'
  or the 'Background' commands, but don't get any sound.
  The logfile says:
  -- Executing BackGround(SIP/joa-9def, tt-weasels) in new stack
  -- Playing 'tt-weasels' (language 'en') Are the sound drivers 
  (alsa or oss) used for this or do I need to configure something else?
  
  thanks for any help,
  Bart
  
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