Justin Killen
2015-Feb-17 01:11 UTC
[asterisk-users] Callfile problem - Unable to find codec translation path from (nothing)
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 at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150216/289c9c24/attachment.html>
Антон Сацкий
2015-Feb-17 11:19 UTC
[asterisk-users] Callfile problem - Unable to find codec translation path from (nothing)
1-wrong AAA/check_ip_failure--- try to use default sounds -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150217/facb1749/attachment.html>
A J Stiles
2015-Feb-17 11:39 UTC
[asterisk-users] Callfile problem - Unable to find codec translation path from (nothing)
On Tuesday 17 Feb 2015, Justin Killen wrote:> 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 .....> 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 .....> Is there something special I need to do to trick the translation into doing > the right thing? > > -JustinYou 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 .
Joshua Colp
2015-Feb-17 12:07 UTC
[asterisk-users] Callfile problem - Unable to find codec translation path from (nothing)
Justin Killen wrote: <snip>> > 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
Justin Killen
2015-Feb-18 20:23 UTC
[asterisk-users] Callfile problem - Unable to find codec translation path from (nothing)
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 at lists.digium.com [mailto:asterisk-users-bounces at 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>> > 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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