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[asterisk-users] Help with voicemail


 
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lucabert at lucabert.de
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PostPosted: Sat Oct 17, 2015 10:13 am    Post subject: [asterisk-users] Help with voicemail Reply with quote

Hi list!

My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a
voicemail.
On two of these numbers the voicemail works without any problem, on the other
it doesn't...
I get this error:

[Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[Oct 17 17:01:29] WARNING[14700]: file.c:957 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/00390151111111/unavail (format 0x100 (g729)): No such file or directory
[Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[Oct 17 17:01:29] WARNING[14700]: file.c:957 ast_streamfile: Unable to open beep (format 0x100 (g729)): No such file or directory
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/00390151111111/tmp/DIqpGh format: wav, 0x6edbd8
-- x=1, open writing: /var/spool/asterisk/voicemail/default/00390151111111/tmp/DIqpGh format: gsm, 0x7c6978
[Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)

Of course, I have a
file /var/spool/asterisk/voicemail/default/00390151111111/unavail.gsm...

Can someone help me to solve my problem?

Thanks a lot!
Luca Bertoncello
(lucabert@lucabert.de)

--
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mjordan at digium.com
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PostPosted: Sat Oct 17, 2015 11:24 am    Post subject: [asterisk-users] Help with voicemail Reply with quote

On Sat, Oct 17, 2015 at 10:12 AM, Luca Bertoncello <lucabert@lucabert.de> wrote:
Quote:
Hi list!

My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a
voicemail.
On two of these numbers the voicemail works without any problem, on the other
it doesn't...
I get this error:

[Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[Oct 17 17:01:29] WARNING[14700]: file.c:957 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/00390151111111/unavail (format 0x100 (g729)): No such file or directory
[Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[Oct 17 17:01:29] WARNING[14700]: file.c:957 ast_streamfile: Unable to open beep (format 0x100 (g729)): No such file or directory
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/00390151111111/tmp/DIqpGh format: wav, 0x6edbd8
-- x=1, open writing: /var/spool/asterisk/voicemail/default/00390151111111/tmp/DIqpGh format: gsm, 0x7c6978
[Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)

Of course, I have a
file /var/spool/asterisk/voicemail/default/00390151111111/unavail.gsm...

Can someone help me to solve my problem?


Do you have a g729 codec module loaded? If so, does it show a
translation path between g729 and gsm? If so, do you have sufficient
encoder/decoder licenses?

Matt

--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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_____________________________________________________________________
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lucabert at lucabert.de
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PostPosted: Sat Oct 17, 2015 11:26 am    Post subject: [asterisk-users] Help with voicemail Reply with quote

Matthew Jordan <mjordan@digium.com> schrieb:

Quote:
Do you have a g729 codec module loaded? If so, does it show a

Bingo!

Quote:
translation path between g729 and gsm? If so, do you have sufficient
encoder/decoder licenses?

I don't have a translation path between g729 and gsm...
Since I don't have a g729 codec, I changed the properties of this peer
enabling other codecs.
Now the voicemail works as expected...

Thanks
Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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sbasan at bluebe.net
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PostPosted: Sat Oct 17, 2015 11:28 am    Post subject: [asterisk-users] Help with voicemail Reply with quote

Check your phone codecs.
It set to g729 while you don't have this codec in your asterisk nor files in this codec. בתאריך 17 באוק' 2015 18:34,‏ "Luca Bertoncello" <lucabert@lucabert.de (lucabert@lucabert.de)> כתב:
Quote:
Hi list!

My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a
voicemail.
On two of these numbers the voicemail works without any problem, on the other
it doesn't...
I get this error:

[Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[Oct 17 17:01:29] WARNING[14700]: file.c:957 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/00390151111111/unavail (format 0x100 (g729)): No such file or directory
[Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[Oct 17 17:01:29] WARNING[14700]: file.c:957 ast_streamfile: Unable to open beep (format 0x100 (g729)): No such file or directory
    -- Recording the message
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/00390151111111/tmp/DIqpGh format: wav, 0x6edbd8
    -- x=1, open writing:  /var/spool/asterisk/voicemail/default/00390151111111/tmp/DIqpGh format: gsm, 0x7c6978
[Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)

Of course, I have a
file /var/spool/asterisk/voicemail/default/00390151111111/unavail.gsm...

Can someone help me to solve my problem?

Thanks a lot!
Luca Bertoncello
(lucabert@lucabert.de (lucabert@lucabert.de))

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
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