Re: [asterisk-users] Sound problem with format files but not codecs

2012-10-22 Thread Administrator TOOTAI 

Le 22/10/2012 04:27, Binan AL Halabi a écrit :

Hello,

It means that one of clients, is using 'silence suppression' mechanism 
which sends audio frames that do not contain any samples.
Asterisk complains about silence supression and appears these warnings 
on  CLI.

If the client turn off the silence suppression the message will disappear.


Hi Binan,

silence suppression is already turned off

Regards



// Binan.
*Från:* Administrator TOOTAI 
*Till:* Asterisk-Users 
*Skickat:* söndag, 21 oktober 2012 10:34
*Ämne:* [asterisk-users] Sound problem with format files but not codecs

Hi all,

on asterisk 1.8.16

[2012-10-20 19:36:17] VERBOSE[743] pbx.c:-- Executing 
[801@OFFICE-Numbers:2] MusicOnHold("Local/801@OFFICE-Numbers-e54a;2", 
"") in new stack
[2012-10-20 19:36:17] VERBOSE[743] res_musiconhold.c:-- Started 
music on hold, class 'TOOTAi', on Local/801@OFFICE-Numbers-e54a;2

[2012-10-20 19:36:17] WARNING[742] translate.c: no samples for ulawtolin
[2012-10-20 19:36:21] VERBOSE[742] pbx.c:  == Spawn extension 
(from_to-OFFICE, 801, 23) exited non-zero on 'SIP/8081773619-2f28'
[2012-10-20 19:36:21] VERBOSE[743] res_musiconhold.c:-- Stopped 
music on hold on Local/801@OFFICE-Numbers-e54a;2


or asterisk 10.8.0

-- Executing [801@macro-GeneralNumbers:1] Set("SIP/105-0081", 
"CHANNEL(musicclass)=TOOTAi") in new stack
-- Executing [801@macro-GeneralNumbers:2] 
MusicOnHold("SIP/105-0081", "") in new stack

-- Started music on hold, class 'TOOTAi', on SIP/105-0081
[2012-10-20 22:48:48] WARNING[22435]: translate.c:343 framein: no 
samples for g722tolin

-- Stopped music on hold on SIP/105-0081

This is when calling extension:

exten=>801,1,Set(CHANNEL(musicclass)=TOOTAi)
exten=>801,n,MusicOnHold()
exten=>801,n,Hangup

What does mean those WARNINGS and how to solve this problem?

MeetMe, Voicemail or holding a call are working fine. From what I 
understand, codecs are used in channels and format for handling files. 
In both cases, two different servers, asterisk is compiled from tar.gz 
and in menuselect all codecs and formats are activated.


Is this a bug? Did I forget something?

On a third server I run latest Elastix with an asterisk 1.8.16 
version. On this server I have no MusicOnHold at all even during 
calls. Logs show


VERBOSE[19717] res_musiconhold.c:-- Started music on hold, class 
'default', on SIP/104-00b3
VERBOSE[19717] res_musiconhold.c:-- Stopped music on hold on 
SIP/104-00b3


which is MusicOnHold stop immediately.

On all servers wav files are installed, even try with original ones 
delivered with Asterisk.


Thanks for any hint


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Re: [asterisk-users] Sound problem with format files but not codecs

2012-10-21 Thread Binan AL Halabi 
Hello,

It means that one of clients, is using 'silence suppression' mechanism 
which sends audio frames that do not contain any samples.
Asterisk complains about silence supression and appears these warnings on  CLI.
If the client turn off the silence suppression the message will disappear.

// Binan.



 Från: Administrator TOOTAI 
Till: Asterisk-Users  
Skickat: söndag, 21 oktober 2012 10:34
Ämne: [asterisk-users] Sound problem with format files but not codecs
 
Hi all,

on asterisk 1.8.16

[2012-10-20 19:36:17] VERBOSE[743] pbx.c:     -- Executing 
[801@OFFICE-Numbers:2] MusicOnHold("Local/801@OFFICE-Numbers-e54a;2", "") in 
new stack
[2012-10-20 19:36:17] VERBOSE[743] res_musiconhold.c:     -- Started music on 
hold, class 'TOOTAi', on Local/801@OFFICE-Numbers-e54a;2
[2012-10-20 19:36:17] WARNING[742] translate.c: no samples for ulawtolin
[2012-10-20 19:36:21] VERBOSE[742] pbx.c:   == Spawn extension (from_to-OFFICE, 
801, 23) exited non-zero on 'SIP/8081773619-2f28'
[2012-10-20 19:36:21] VERBOSE[743] res_musiconhold.c:     -- Stopped music on 
hold on Local/801@OFFICE-Numbers-e54a;2

or asterisk 10.8.0

    -- Executing [801@macro-GeneralNumbers:1] Set("SIP/105-0081", 
"CHANNEL(musicclass)=TOOTAi") in new stack
    -- Executing [801@macro-GeneralNumbers:2] MusicOnHold("SIP/105-0081", 
"") in new stack
    -- Started music on hold, class 'TOOTAi', on SIP/105-0081
[2012-10-20 22:48:48] WARNING[22435]: translate.c:343 framein: no samples for 
g722tolin
    -- Stopped music on hold on SIP/105-0081

This is when calling extension:

exten=>801,1,Set(CHANNEL(musicclass)=TOOTAi)
exten=>801,n,MusicOnHold()
exten=>801,n,Hangup

What does mean those WARNINGS and how to solve this problem?

MeetMe, Voicemail or holding a call are working fine. From what I understand, 
codecs are used in channels and format for handling files. In both cases, two 
different servers, asterisk is compiled from tar.gz and in menuselect all 
codecs and formats are activated.

Is this a bug? Did I forget something?

On a third server I run latest Elastix with an asterisk 1.8.16 version. On this 
server I have no MusicOnHold at all even during calls. Logs show

VERBOSE[19717] res_musiconhold.c:     -- Started music on hold, class 
'default', on SIP/104-00b3
VERBOSE[19717] res_musiconhold.c:     -- Stopped music on hold on 
SIP/104-00b3

which is MusicOnHold stop immediately.

On all servers wav files are installed, even try with original ones delivered 
with Asterisk.

Thanks for any hint

Regards
-- Daniel

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[asterisk-users] Sound problem with format files but not codecs

2012-10-21 Thread Administrator TOOTAI 

Hi all,

on asterisk 1.8.16

[2012-10-20 19:36:17] VERBOSE[743] pbx.c: -- Executing 
[801@OFFICE-Numbers:2] MusicOnHold("Local/801@OFFICE-Numbers-e54a;2", 
"") in new stack
[2012-10-20 19:36:17] VERBOSE[743] res_musiconhold.c: -- Started 
music on hold, class 'TOOTAi', on Local/801@OFFICE-Numbers-e54a;2

[2012-10-20 19:36:17] WARNING[742] translate.c: no samples for ulawtolin
[2012-10-20 19:36:21] VERBOSE[742] pbx.c:   == Spawn extension 
(from_to-OFFICE, 801, 23) exited non-zero on 'SIP/8081773619-2f28'
[2012-10-20 19:36:21] VERBOSE[743] res_musiconhold.c: -- Stopped 
music on hold on Local/801@OFFICE-Numbers-e54a;2


or asterisk 10.8.0

-- Executing [801@macro-GeneralNumbers:1] Set("SIP/105-0081", 
"CHANNEL(musicclass)=TOOTAi") in new stack
-- Executing [801@macro-GeneralNumbers:2] 
MusicOnHold("SIP/105-0081", "") in new stack

-- Started music on hold, class 'TOOTAi', on SIP/105-0081
[2012-10-20 22:48:48] WARNING[22435]: translate.c:343 framein: no 
samples for g722tolin

-- Stopped music on hold on SIP/105-0081

This is when calling extension:

exten=>801,1,Set(CHANNEL(musicclass)=TOOTAi)
exten=>801,n,MusicOnHold()
exten=>801,n,Hangup

What does mean those WARNINGS and how to solve this problem?

MeetMe, Voicemail or holding a call are working fine. From what I 
understand, codecs are used in channels and format for handling files. 
In both cases, two different servers, asterisk is compiled from tar.gz 
and in menuselect all codecs and formats are activated.


Is this a bug? Did I forget something?

On a third server I run latest Elastix with an asterisk 1.8.16 version. 
On this server I have no MusicOnHold at all even during calls. Logs show


VERBOSE[19717] res_musiconhold.c: -- Started music on hold, class 
'default', on SIP/104-00b3
VERBOSE[19717] res_musiconhold.c: -- Stopped music on hold on 
SIP/104-00b3


which is MusicOnHold stop immediately.

On all servers wav files are installed, even try with original ones 
delivered with Asterisk.


Thanks for any hint

Regards
--
Daniel

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Re: [Asterisk-Users] Sound problem

2005-12-05 Thread Giovanni Miano 
Attention:Your mp3s arent higher than 128 bit/s2005/12/5, Vipul Patel <[EMAIL PROTECTED]>:
Hi all

I had allread install asterisk server and two X-Lite softphones on two
different machines. whole processa of calling is going fine. But I
cann't able to hear ringing / any type of voice on both side.

The asterisk sever give following worning.

WARNING[1922]: res_musiconhold.c:205 spawn_mp3: Found no files in '/usr/share/asterisk/mohmp3'
Dec  5 15:41:27 WARNING[1922]: res_musiconhold.c:278 monmp3thread: unable to spawn mp3player

Can any one help? what is wrong with this.

Thanks
Vipul

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[Asterisk-Users] Sound problem

2005-12-05 Thread Vipul Patel 
Hi all

I had allread install asterisk server and two X-Lite softphones on two
different machines. whole processa of calling is going fine. But I
cann't able to hear ringing / any type of voice on both side.

The asterisk sever give following worning.

WARNING[1922]: res_musiconhold.c:205 spawn_mp3: Found no files in '/usr/share/asterisk/mohmp3'
Dec  5 15:41:27 WARNING[1922]: res_musiconhold.c:278 monmp3thread: unable to spawn mp3player

Can any one help? what is wrong with this.

Thanks
Vipul
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Re: [Asterisk-Users] sound problem, please help!

2005-11-28 Thread Esteban Maestre 
Hello, Rusty!

Thanks for your reply... You have been the only one ;)

Hahaha, it could become dangerous... ;)

Well, I have been investigating it a little bit, and I guess the main
reason is something related to what you have pointed in the first
paragraph.
I've found some interesting information here:

http://www.privateline.com/PCS/GSM08.html

There is an interesting article I've already tried to post, but toy can
find it easily in the www above.

regards,
-esteban-


> I don't think you'd want to hear most of what I say when I am on hold,
> especially if I am on hold with a telephone company! It tends to be
> somewhat
> on the profane side. *grin*
>
> -Rusty
>
> On 11/25/05, Esteban Maestre <[EMAIL PROTECTED]> wrote:
>>
>> kind of... ;)
>> I want to know what the people say when they are waiting... :P
>>
>> do you have any idea on what the problem could be?
>>
>> -esteban-
>>
>>
>> >  Original Message 
>> > From: "Esteban Maestre" <[EMAIL PROTECTED]>
>> > To: 
>> > Sent: Friday, November 25, 2005 11:22 AM
>> > Subject: [Asterisk-Users] sound problem, please help!
>> >
>> >> Hi all!
>> >>
>> >> I have a strange problem when using asterisk. I have configured
>> >> asterisk to receive calls (FX0). In my configuration, I want asterisk
>> >> to play music while  I record the caller's speech.
>> >
>> > Dialup-karaoke? :-)
>> >
>> > Leif
>> >
>> >
>> >
>> >
>>
>>
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>>
>


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Re: [Asterisk-Users] sound problem, please help!

2005-11-28 Thread Rusty Dekema 
Hi,

I have noticed that most mobile phones (GSM and CDMA at least) seem to
have a tendency to interrupt the incoming audio stream when the
microphone levels get louder than a certain threshold (such as when you
are speaking into it). I do not know exactly why this happens, nor
whether it is something that happens in the handset (the handset mutes
the incoming audio, which doesn't make much sense) or in the network
(maybe to try to save bandwidth?). 
Also, the GSM codec is intended to intelligibly encode human speech,
not general audio signals such as music. GSM uses approximately 6-8
times less bandwidth than G.711u/a, and the degradation in the quality
of music and other non-speech signals is one of the tradeoffs that we
make in order to not have the cost of mobile calls increase 6-8 times
over.

Maybe if Europe was as backward with its mobile systems as the United
States currently is, you could try to get your users to call your
system from analog mobile phones... They don't use digital compression
(by definition, I suppose) and would probably work much better for your
specific application ;).

-Rusty


On 11/25/05, Esteban Maestre <[EMAIL PROTECTED]> wrote:
Hi all!I have a strange problem when using asterisk. I have configured asteriskto receive calls (FX0). In my configuration, I want asterisk to play musicwhile  I record the caller's speech. If the caller does the call from a
fixed line telephone, there is no problem, but in case the caller does thecall from a mobile GSM phone, the quality of the music he hears becomes sobad, and even more when he speaks.I have tried several codecs.
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Re: [Asterisk-Users] sound problem, please help!

2005-11-25 Thread Esteban Maestre 
kind of... ;)
I want to know what the people say when they are waiting... :P

do you have any idea on what the problem could be?

-esteban-


>  Original Message 
> From: "Esteban Maestre" <[EMAIL PROTECTED]>
> To: 
> Sent: Friday, November 25, 2005 11:22 AM
> Subject: [Asterisk-Users] sound problem, please help!
>
>> Hi all!
>>
>> I have a strange problem when using asterisk. I have configured
>> asterisk to receive calls (FX0). In my configuration, I want asterisk
>> to play music while  I record the caller's speech.
>
> Dialup-karaoke? :-)
>
> Leif
>
>
>
>


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Re: [Asterisk-Users] sound problem, please help!

2005-11-25 Thread Leif Neland 

 Original Message 
From: "Esteban Maestre" <[EMAIL PROTECTED]>
To: 
Sent: Friday, November 25, 2005 11:22 AM
Subject: [Asterisk-Users] sound problem, please help!


Hi all!

I have a strange problem when using asterisk. I have configured
asterisk to receive calls (FX0). In my configuration, I want asterisk
to play music while  I record the caller's speech.


Dialup-karaoke? :-)

Leif

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[Asterisk-Users] sound problem, please help!

2005-11-25 Thread Esteban Maestre 
Hi all!

I have a strange problem when using asterisk. I have configured asterisk
to receive calls (FX0). In my configuration, I want asterisk to play music
while  I record the caller's speech. If the caller does the call from a
fixed line telephone, there is no problem, but in case the caller does the
call from a mobile GSM phone, the quality of the music he hears becomes so
bad, and even more when he speaks.
I have tried several codecs.
Any idea or advice?

thanks in advance,

-esteban-

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Re: [Asterisk-Users] Sound Problem

2005-02-12 Thread Jon Gabrielson 
You need to tell us what type of device you are
using to make the phone calls.  Are you using
a ZAP FXS, a softphone, a sip phone, or an iax phone.
Also, how are you terminating the call.  Is it via a
ZAP FXO device like a t100p, is it another VOIP phone,
or is it via a service provider like iax.cc or nufone?

Cheers,


Jon.


On Saturday 12 February 2005 02:20 pm, chawki hammoud wrote:
> I have been using Asterisk to make phone calls and
> when i tried to use it today the volume was
> unexpectedly very low. Changing the volume in the
> volume control didn't effect it. I believe the problem
> lies with Asterisk and not the volume control. I
> appreciate any feedback of where and what to check.
>
>
>
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[Asterisk-Users] Sound Problem

2005-02-12 Thread chawki hammoud 
I have been using Asterisk to make phone calls and
when i tried to use it today the volume was
unexpectedly very low. Changing the volume in the
volume control didn't effect it. I believe the problem
lies with Asterisk and not the volume control. I
appreciate any feedback of where and what to check. 



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RE : [Asterisk-Users] sound problem

2005-01-11 Thread Jalil BOUREKBA 
Check the used codecs, if it´s according to the codec used by your sip
users.

Welcome

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Muhammad
Rizwan Khan
Envoyé : lundi 10 janvier 2005 18:53
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] sound problem


Is there any config file related to voice, which should be change in
order to 
hear the sound in dialer?

On Monday 10 January 2005 21:23, you wrote:
> I have configured asterisk, but when i calls from my dialler, it
connects
> successfully, but did not give any voice at both ends.
> What should i need to do?
>
> Thanks
>
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Re: [Asterisk-Users] sound problem

2005-01-10 Thread Muhammad Rizwan Khan 

Is there any config file related to voice, which should be change in order to 
hear the sound in dialer?

On Monday 10 January 2005 21:23, you wrote:
> I have configured asterisk, but when i calls from my dialler, it connects
> successfully, but did not give any voice at both ends.
> What should i need to do?
>
> Thanks
>
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[Asterisk-Users] sound problem

2005-01-10 Thread Muhammad Rizwan Khan 
I have configured asterisk, but when i calls from my dialler, it connects 
successfully, but did not give any voice at both ends.
What should i need to do?

Thanks

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[Asterisk-Users] Sound Problem

2004-07-30 Thread Sascha Growe 
Hello there,

Ich have some interesint problems with SIP & CAPI.
Im routing incoming SIP Calls through CAPI to a telephone @ work, but I cant
get any sound.
If I call from sip client to CAPI direktly I have sound.
If I'm recording to a wave file with asterisk there is some sound.

My SIP Provider is sipgate

Sascha Growe

PS.: Sorry for my bad english.

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[Asterisk-Users] Sound Problem

2004-04-22 Thread Dan 
I have some incoming did`s from a sip provider..  The problem i`m having is when
no one is talking it sounds like someone is faxing or sending Data.

I can`t figure out for the life of me what the problem is  I`m not receiving any
errors from AsteriskCould someone please direct me to a way to fix this


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RE: [Asterisk-Users] sound problem

2003-08-18 Thread Wade Weppler 
Most OSS drivers don't support full duplex.  We've upgraded to ALSA, and
most problems disappear.

-wade

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of santiago
> Sent: Monday, August 18, 2003 10:36 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] sound problem
> 
> hi list,
> 
> when I run asterisk, appears the following:
> 
> 
> WARNING[1074459808]: File chan_oss.c, Line 346 (setformat): Requested
> 8000 Hz, got 8178 Hz -- sound may be choppy
> WARNING[1074459808]: File chan_oss.c, Line 974 (load_module): XXX I
> don't work right with non-full duplex sound cards XXX
> WARNING[1133735216]: File chan_oss.c, Line 232 (sound_thread): Read
> error on sound device: Resource temporarily unavailable
> 
> 
> 
> but I can use oss with xmms
> 
> what i have to do?
> 
> thanks,
> 
> 
> --
> santiago josé ruano rincón
> administración servidores y servicios de internet
> red de datos
> universidad del cauca
> 
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> Version: GnuPG v1.0.6 (GNU/Linux)
> Comment: For info see http://www.gnupg.org
> 
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> zMtXCMrMS/6cxxWal1mWWlScmZKYopCSmqPgnFianMjF1WHPzMoA0gozUJDpLSfD
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> 
> 
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> Asterisk-Users mailing list
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[Asterisk-Users] sound problem

2003-08-18 Thread santiago 
hi list,

when I run asterisk, appears the following:


WARNING[1074459808]: File chan_oss.c, Line 346 (setformat): Requested
8000 Hz, got 8178 Hz -- sound may be choppy
WARNING[1074459808]: File chan_oss.c, Line 974 (load_module): XXX I
don't work right with non-full duplex sound cards XXX
WARNING[1133735216]: File chan_oss.c, Line 232 (sound_thread): Read
error on sound device: Resource temporarily unavailable



but I can use oss with xmms

what i have to do?

thanks,


-- 
santiago josé ruano rincón
administración servidores y servicios de internet
red de datos
universidad del cauca
 
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Version: GnuPG v1.0.6 (GNU/Linux)
Comment: For info see http://www.gnupg.org
 
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/IB9Gre8ZHfZ+/BvkX4osko5wPfUQ4b5ST8VT3hciP3inJl578O17DedDS9fAgA=
=5oc0
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