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 at lucabert.de)
On Sat, Oct 17, 2015 at 10:12 AM, Luca Bertoncello <lucabert at lucabert.de> wrote:> 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
Matthew Jordan <mjordan at digium.com> schrieb:> Do you have a g729 codec module loaded? If so, does it show aBingo!> 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 at lucabert.de)
Check your phone codecs. It set to g729 while you don't have this codec in your asterisk nor files in this codec. ?????? 17 ????' 2015 18:34,? "Luca Bertoncello" <lucabert at lucabert.de> ???:> 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 at 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151017/f8cfadf4/attachment.html>