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jkillen at allamerican...
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 Posted: Mon Feb 16, 2015 8:12 pm Post subject: [asterisk-users] Callfile problem - Unable to find codec tra |
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Hi,
I copied a setup from an older 1.8.5 installation to an 11.15 installation, and I’m having problems getting call files to work. Here is the extension setup I’m using:
[outbound-swift]
exten => _[a-zA-Z].,1,Answer
exten => _[a-zA-Z].,n,Playback(AAA/check_ip_failure)
;exten => _[a-zA-Z].,1,Swift("${EXTEN}")
exten => _[a-zA-Z].,n,Goto(1)
[mis-phone]
exten => _X.,n,Dial(DAHDI/g51/w${EXTEN},15,r)
exten => _X.,n,Hangup
and here is a sample call file:
Channel: Local/5551212@mis-phone/n
MaxRetries: 50
RetryTime: 5
WaitTime: 5
Archive: yes
Extension: ernestine ip monitor failure for dozer ping
Context: outbound-swift
Whenever I try to copy this callfile into /var/spool/asterisk/outgoing/ I get these 3 lines repeating over and over (I’m not 100% sure which entry is first):
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: channel.c:5353 set_format: Unable to find a codec translation path from (nothing) to (slin)
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: file.c:1017 ast_streamfile: Unable to open AAA/check_ip_failure (format (nothing)): Function not implemented
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: app_playback.c:484 playback_exec: ast_streamfile failed on OutgoingSpoolFailed for AAA/check_ip_failure
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: channel.c:5353 set_format: Unable to find a codec translation path from (nothing) to (slin)
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: file.c:1017 ast_streamfile: Unable to open AAA/check_ip_failure (format (nothing)): Function not implemented
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: app_playback.c:484 playback_exec: ast_streamfile failed on OutgoingSpoolFailed for AAA/check_ip_failure
Is there something special I need to do to trick the translation into doing the right thing?
-Justin |
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satskiy.a at gmail.com
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| Posted: Tue Feb 17, 2015 6:19 am Post subject: [asterisk-users] Callfile problem - Unable to find codec tra |
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| 1-wrong AAA/check_ip_failure--- try to use default sounds |
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asterisk_list at earth...
Guest
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| Posted: Tue Feb 17, 2015 6:40 am Post subject: [asterisk-users] Callfile problem - Unable to find codec tra |
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On Tuesday 17 Feb 2015, Justin Killen wrote:
| Quote: | Hi,
I copied a setup from an older 1.8.5 installation to an 11.15 installation,
and I'm having problems getting call files to work.
|
..... stuff deleted .....
| Quote: | Whenever I try to copy this callfile into /var/spool/asterisk/outgoing/ I
get these 3 lines repeating over and over (I'm not 100% sure which entry
is first):
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: channel.c:5353 set_format:
Unable to find a codec translation path from (nothing) to (slin)
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: file.c:1017
ast_streamfile: Unable to open AAA/check_ip_failure (format (nothing)):
Function not implemented [2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]:
app_playback.c:484 playback_exec: ast_streamfile failed on
OutgoingSpoolFailed for AAA/check_ip_failure [2015-02-16 16:56:02]
|
/\ /\ /\ /\ /\ /\ THIS IS THE PROBLEM /\ /\ /\ /\ /\
..... stuff deleted .....
| Quote: | Is there something special I need to do to trick the translation into doing
the right thing?
-Justin
|
You need to have the sound file saved in the correct place, and Asterisk has to
be able to read it. Double-check, triple-check and check for a fourth time
that the file is really where you think it is and its permissions, and those of
the containing folder, are correct.
Whenever I need to use a custom sound file or files, then I usually set up a
test extension which just plays my wanted sound file(s) and calls Hangup() .
Then I call this and test that my custom sounds work.
--
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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To UNSUBSCRIBE or update options visit:
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jcolp at digium.com
Guest
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| Posted: Tue Feb 17, 2015 7:07 am Post subject: [asterisk-users] Callfile problem - Unable to find codec tra |
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Justin Killen wrote:
<snip>
| Quote: |
Whenever I try to copy this callfile into /var/spool/asterisk/outgoing/
I get these 3 lines repeating over and over (I’m not 100% sure which
entry is first):
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: channel.c:5353
set_format: Unable to find a codec translation path from (nothing) to (slin)
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: file.c:1017
ast_streamfile: Unable to open AAA/check_ip_failure (format (nothing)):
Function not implemented
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: app_playback.c:484
playback_exec: ast_streamfile failed on OutgoingSpoolFailed for
AAA/check_ip_failure
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: channel.c:5353
set_format: Unable to find a codec translation path from (nothing) to (slin)
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: file.c:1017
ast_streamfile: Unable to open AAA/check_ip_failure (format (nothing)):
Function not implemented
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: app_playback.c:484
playback_exec: ast_streamfile failed on OutgoingSpoolFailed for
AAA/check_ip_failure
Is there something special I need to do to trick the translation into
doing the right thing?
|
It can never do the right thing there. If the origination fails for some
reason then a channel (without any formats) is created to the
"OutgoingSpoolFailed" extension. Due to the way you've written your
dialplan logic this will attempt to do things with media. Since it's not
a real channel and has no formats, that will fail. Since your dialplan
logic also has it go in a loop it just goes 'round and 'round.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- 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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jkillen at allamerican...
Guest
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| Posted: Wed Feb 18, 2015 3:24 pm Post subject: [asterisk-users] Callfile problem - Unable to find codec tra |
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Joshua,
If I'm understanding this correctly, you're saying that the Playback is failing because it isn't connected to anything on the other end, because the Dial() failed. When the channel is created on the "OutgoingSpoolFailed" extension, what context is it created in, one of the origin legs? Is there a way detect this condition in the target context ([outbound-swift]), or better yet, verify the other leg is attached before starting the logic?
-Justin
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Joshua Colp
Sent: Tuesday, February 17, 2015 4:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Callfile problem - Unable to find codec translation path from (nothing)
Justin Killen wrote:
<snip>
| Quote: |
Whenever I try to copy this callfile into
/var/spool/asterisk/outgoing/ I get these 3 lines repeating over and
over (I'm not 100% sure which entry is first):
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: channel.c:5353
set_format: Unable to find a codec translation path from (nothing) to
(slin)
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: file.c:1017
ast_streamfile: Unable to open AAA/check_ip_failure (format (nothing)):
Function not implemented
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: app_playback.c:484
playback_exec: ast_streamfile failed on OutgoingSpoolFailed for
AAA/check_ip_failure
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: channel.c:5353
set_format: Unable to find a codec translation path from (nothing) to
(slin)
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: file.c:1017
ast_streamfile: Unable to open AAA/check_ip_failure (format (nothing)):
Function not implemented
[2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: app_playback.c:484
playback_exec: ast_streamfile failed on OutgoingSpoolFailed for
AAA/check_ip_failure
Is there something special I need to do to trick the translation into
doing the right thing?
|
It can never do the right thing there. If the origination fails for some reason then a channel (without any formats) is created to the "OutgoingSpoolFailed" extension. Due to the way you've written your dialplan logic this will attempt to do things with media. Since it's not a real channel and has no formats, that will fail. Since your dialplan logic also has it go in a loop it just goes 'round and 'round.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- 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
--
_____________________________________________________________________
-- 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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jcolp at digium.com
Guest
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| Posted: Wed Feb 18, 2015 3:32 pm Post subject: [asterisk-users] Callfile problem - Unable to find codec tra |
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Justin Killen wrote:
| Quote: | Joshua,
If I'm understanding this correctly, you're saying that the Playback
is failing because it isn't connected to anything on the other end,
because the Dial() failed. When the channel is created on the
"OutgoingSpoolFailed" extension, what context is it created in, one
of the origin legs? Is there a way detect this condition in the
target context ([outbound-swift]), or better yet, verify the other
leg is attached before starting the logic?
|
It is created in the context you have told the answered channel to go
to. One way to fix it would be to add an OutgoingSpoolFailed extension
for each priority so that Swift isn't invoked. You could also use GotoIf
in the first priority and go elsewhere if the extension is
OutgoingSpoolFailed. There exist many options.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- 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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