[asterisk-users] Mp3player
hi all - has anyone used gstreamer to have and audio source (some music playing on a PC for example) and then use Mp3Player() to play that source ? what is the gstreamer command to do that ? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MP3Player audio format
Hi, I've successfully installed Asterisk and placed test calls using the MP3Player application. However I notice that my call quality varies drastically depending on which MP3 I use. Since I'm not changing any settings I'm assuming that the encoding of the file makes a difference. I've tried with both the g729 and alaw codec and they both give the same replicable results. I read that the underlying program mpg123 prefers mp3 files without ID3 tags. What is the recommended audio format (bit/sec) for using the MP3Player application or does this not make a difference? Thanks, Andy -- _ -- 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
[asterisk-users] mp3player() to shuffle playlist
Hi all, I am trying to make a scenario when someone dial *10*, the mp3player() function would act and play a list of MP3 files. However, I have no idea how to randomize the function (mpg123 is capable of shuffling the MP3 files, buat how to implement it in extensions.conf?) Perhaps any of you sucessfully made a playlist and shuffling method for mp3player() in extensions.conf and would like to share it with me. thanks all for your comments ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MP3Player
Hi All, I'm having problem with MP3Player app. I want the caller to hear mp3 when he is waiting until I answer my phone. -- from extentions.conf -- exten = 200,1,Answer() exten = 200,2,MP3Player(/home/user200/mp3/hanna-hais.mp3) exten = 200,3,Dial(SIP/200|20|tTrR) exten = 200,4,Hangup() -- end -- here is debug from CLI: -- Executing Answer(SIP/200-08a64d98, ) in new stack -- Executing MP3Player(SIP/200-08a64d98, /home/user200/mp3/hanna-hais.mp3) in new stack Mar 15 11:25:32 NOTICE[4991]: app_mp3.c:121 timed_read: Poll timed out/errored out with 0 -- Executing Dial(SIP/200-08a64d98, SIP/200|20|tTrR) in new stack -- Called 200 -- SIP/200-08a6a2d8 is ringing Asterisk 1.2.16 and mpg123 installed. Any ideas? Thank you in advance, Dominik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MP3player distortion with Asterisk 1.4
I upgraded my Asterisk configuration to 1.4.0 yesterday, when I was adding a TDM400P to have two PSTN connections to my analog phone lines. Adding the phone lines was a success, however, I now notice the MP3Player audio sounds horrible (incomprehensible). The only changes that I have made to the environment were the upgrade from 1.2 to 1.4. Any thoughts? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] MP3player Problem
-Ursprüngliche Nachricht- Von: Bayrouni [mailto:[EMAIL PROTECTED] Gesendet: Dienstag, 7. Februar 2006 21:39 An: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] MP3player Problem office wrote: Hi, i use in my extensions.conf a testline for an internal test : exten = 10,1,MP3Player(/var/lib/asterisk/mohmp3/fpm-calm-river.mp3) When i call 10, Asterisk answer and i see in the CLI, that MP3player works without problems - but i can't hear the sound at the phone ? Where is the Problem ? Walter Hello, you are missing mpg123,Install it and moh will work -- Bayrouni Hello, I have already installed mpg123 and it works fine ! When i start mpg123 i see these correct outputs : High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp. Uses code from various people. See 'README' for more! THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK! This ist the CLI-Output, when i try an internal call to 10 : *CLI == Primary D-Channel on span 1 up for TEI 64 -- Accepting overlap voice call from '11' to '10' on channel 0/2, span 1 -- Starting simple switch on 'Zap/2-1' -- Executing Set(Zap/2-1, LANGUAGE()=de) in new stack -- Executing Answer(Zap/2-1, ) in new stack -- Executing MP3Player(Zap/2-1, /var/lib/asterisk/mohmp3/fpm-calm-river.mp3) in new stack Now i don't know, which job make asterisk in the background - but after 2min. Asterisk hangup with this : Feb 8 15:56:59 NOTICE[29435]: app_mp3.c:108 timed_read: Poll timed out/errored out with 0 -- Executing PlayTones(Zap/2-1, congestion) in new stack -- Executing Hangup(Zap/2-1, ) in new stack == Spawn extension (zap-incoming, 10, 5) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' -- Executing PlayTones(Zap/2-1, congestion) in new stack Walter ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] MP3player Problem
Just a long shot: Do you need to Answer() the call first? exten = 10,1,Answer exten = 10,2,MP3Player(...) Like this. Rene Kluwen Chimit -Ursprüngliche Nachricht- Von: Bayrouni [mailto:[EMAIL PROTECTED] Gesendet: Dienstag, 7. Februar 2006 21:39 An: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] MP3player Problem office wrote: Hi, i use in my extensions.conf a testline for an internal test : exten = 10,1,MP3Player(/var/lib/asterisk/mohmp3/fpm-calm-river.mp3) When i call 10, Asterisk answer and i see in the CLI, that MP3player works without problems - but i can't hear the sound at the phone ? Where is the Problem ? Walter Hello, you are missing mpg123,Install it and moh will work -- Bayrouni Hello, I have already installed mpg123 and it works fine ! When i start mpg123 i see these correct outputs : High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp. Uses code from various people. See 'README' for more! THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK! This ist the CLI-Output, when i try an internal call to 10 : *CLI == Primary D-Channel on span 1 up for TEI 64 -- Accepting overlap voice call from '11' to '10' on channel 0/2, span 1 -- Starting simple switch on 'Zap/2-1' -- Executing Set(Zap/2-1, LANGUAGE()=de) in new stack -- Executing Answer(Zap/2-1, ) in new stack -- Executing MP3Player(Zap/2-1, /var/lib/asterisk/mohmp3/fpm-calm-river.mp3) in new stack Now i don't know, which job make asterisk in the background - but after 2min. Asterisk hangup with this : Feb 8 15:56:59 NOTICE[29435]: app_mp3.c:108 timed_read: Poll timed out/errored out with 0 -- Executing PlayTones(Zap/2-1, congestion) in new stack -- Executing Hangup(Zap/2-1, ) in new stack == Spawn extension (zap-incoming, 10, 5) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' -- Executing PlayTones(Zap/2-1, congestion) in new stack Walter ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: AW: [Asterisk-Users] MP3player Problem
Just a long shot: Do you need to Answer() the call first? exten = 10,1,Answer exten = 10,2,MP3Player(...) Hello Rene This is already in extensions.conf See below: -- Starting simple switch on 'Zap/2-1' -- Executing Set(Zap/2-1, LANGUAGE()=de) in new stack -- Executing Answer(Zap/2-1, ) in new stack -- Executing MP3Player(Zap/2-1,/var/lib/asterisk/mohmp3/fpm-calm-river.mp3) in new stack Like this. Rene Kluwen Chimit -Ursprüngliche Nachricht- Von: Bayrouni [mailto:[EMAIL PROTECTED] Gesendet: Dienstag, 7. Februar 2006 21:39 An: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] MP3player Problem office wrote: Hi, i use in my extensions.conf a testline for an internal test : exten = 10,1,MP3Player(/var/lib/asterisk/mohmp3/fpm-calm-river.mp3) When i call 10, Asterisk answer and i see in the CLI, that MP3player works without problems - but i can't hear the sound at the phone ? Where is the Problem ? Walter Hello, you are missing mpg123,Install it and moh will work -- Bayrouni Hello, I have already installed mpg123 and it works fine ! When i start mpg123 i see these correct outputs : High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp. Uses code from various people. See 'README' for more! THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK! This ist the CLI-Output, when i try an internal call to 10 : *CLI == Primary D-Channel on span 1 up for TEI 64 -- Accepting overlap voice call from '11' to '10' on channel 0/2, span 1 -- Starting simple switch on 'Zap/2-1' -- Executing Set(Zap/2-1, LANGUAGE()=de) in new stack -- Executing Answer(Zap/2-1, ) in new stack -- Executing MP3Player(Zap/2-1, /var/lib/asterisk/mohmp3/fpm-calm-river.mp3) in new stack Now i don't know, which job make asterisk in the background - but after 2min. Asterisk hangup with this : Feb 8 15:56:59 NOTICE[29435]: app_mp3.c:108 timed_read: Poll timed out/errored out with 0 -- Executing PlayTones(Zap/2-1, congestion) in new stack -- Executing Hangup(Zap/2-1, ) in new stack == Spawn extension (zap-incoming, 10, 5) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' -- Executing PlayTones(Zap/2-1, congestion) in new stack Walter ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MP3player Problem
Hi, i use in my extensions.conf a testline for an internal test : exten = 10,1,MP3Player(/var/lib/asterisk/mohmp3/fpm-calm-river.mp3) When i call 10, Asterisk answer and i see in the CLI, that MP3player works without problems - but i can't hearthe soundat the phone ? Where is the Problem ? Walter ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3player Problem
office wrote: Hi, i use in my extensions.conf a testline for an internal test : exten = 10,1,MP3Player(/var/lib/asterisk/mohmp3/fpm-calm-river.mp3) When i call 10, Asterisk answer and i see in the CLI, that MP3player works without problems - but i can't hear the sound at the phone ? Where is the Problem ? Walter Hello, you are missing mpg123,Install it and moh will work -- Bayrouni ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MP3PLayer - rewind, forward, pause functions - feature request
mpg123 has these undocumented functions. start mpg123 with a -C argument and then pressing s will stop(pause toggle), pressing . will forward and pressing , will rewind. Can these be added to the MP3Player feature? I am using asterisk to play podcasts and this would be a nice feature. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MP3Player cmd issue
I am running CVS HEAD (on a Linux-PPC machine.) My current dialplan generates an error at the console in asterisk when I attempt to issue the MP3Player command -- I can't figure out why it's not playing the actual audio file? The rest of the dialplan works great. Here's what I see in the console: -- Executing MP3Player(IAX2/[EMAIL PROTECTED], /private/var/lib/asterisk/sounds/audiofile.mp3) in new stack Aug 18 15:04:19 NOTICE[17387]: chan_iax2.c:3006 iax2_read: I should never be called! Aug 18 15:04:22 NOTICE[17387]: app_mp3.c:96 timed_read: Poll timed out/errored out with 0 Does anyone have any ideas what's going on or where I could look to trouble shoot this further? MPG123 plays the audio file fine at the command line, just not thru asterisk. And this same dialplan works fine under CVS HEAD, causing MPG123 to play the audio file fine, on OSX so it's something funky and I am not sure what it is. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MP3Player could not play remote stream
Hi all, I could use MP3Player to play local sound (e.g: /usr/sound/abc.mp3) but I could not use it to run a remote stream, if I use mpg123 in command line, I can hear the audio ( /usr/bin/mpg123 http://...), but the same remote mp3 file could not be replay with asterisk. I would appreciate with any suggestion. Phuong Here are the log message: -- Starting simple switch on 'Zap/3-1' Jun 2 16:28:46 NOTICE[3129]: chan_zap.c:5374 ss_thread: Got event 2 (Ring/Answered)... -- Executing Answer(Zap/3-1, ) in new stack -- Executing MP3Player(Zap/3-1, http://localhost:8000/;) in new stack Jun 2 16:28:57 NOTICE[3129]: app_mp3.c:91 timed_read: Poll timed out/errored out with 0 -- Executing Goto(Zap/3-1, mainmenu|s|1) in new stack -- Goto (mainmenu,s,1) -- Executing Answer(Zap/3-1, ) in new stack -- Executing MP3Player(Zap/3-1, http://localhost:8000/;) in new stack extension.conf: (URL is point to a stream server and it works with command line mpg123!) exten = s,2,MP3Player(http://localhost:8000/) -- Geschenkt: 3 Monate GMX ProMail gratis + 3 Ausgaben stern gratis ++ Jetzt anmelden testen ++ http://www.gmx.net/de/go/promail ++ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mp3player - sounds terrible
I tried for the first time to use mp3player(toto.mp3) in my extensions.conf. This part works perfectly. But the sound quality is terrible. I tried to play the mp3 with mpg123 on my audio device, plays ok. And I cannot found any information about the mp3player application. Do I need to encode the mp3 at a particular bitrate ? Does someone had this problem before ? -Yves ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MP3Player strange error
Hi all! I downloaded right mpg123, chabged path to mpg123 binary in app_mp3.c, rebuilt app_mp3.so, and got MusicOnHold to work. But MP3Player refuses to do properly: -- Accepting AUTHENTICATED call from x.x.x.x, requested format = 1024, actual format = 1024 -- Executing Answer(IAX2/[EMAIL PROTECTED]/3, ) in new stack -- Executing MP3Player(IAX2/[EMAIL PROTECTED]/3, mohmp3) in new stack Aug 31 08:07:25 NOTICE[1135618864]: chan_iax2.c:2375 iax2_read: I should never be called! Aug 31 08:07:28 NOTICE[1135618864]: app_mp3.c:91 timed_read: Selected timed out/errored out with 0 -- Timeout on IAX2/[EMAIL PROTECTED]/3 == CDR updated on IAX2/[EMAIL PROTECTED]/3 -- Executing Hangup(IAX2/[EMAIL PROTECTED]/3, ) in new stack == Spawn extension (litnimax-in, t, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/3' -- Executing Hangup(IAX2/[EMAIL PROTECTED]/3, ) in new stack == Spawn extension (litnimax-in, h, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/3' -- Hungup 'IAX2/[EMAIL PROTECTED]/3' new*CLI ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MP3Player problems...
Okay, I've looked around for a FAQ on this, and it's kinda driving me up the wall. I recently installed a new Asterisk system, and built mpg123 v. 0.59s from sources, and the audio comes out wy overmodulated. Any thoughts? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mp3player not working
Problem solved found that Asterisk is calling mpg123 to playback mp3s which isnt installed on Slackware 9.1 by default. Downloaded mpg123 source from http://www.mpg123.de/ and compiled with make linux; make install and now working. Also discovered that mpg123 doesnt seem to playback mp3s with ID3 tags in them, so strip them out before copying them to your Asterisk box. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Lee Sent: Thursday, 22 January 2004 11:18 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] mp3player not working Hi, Im running the latest Asterisk (built last Saturday) and cant get mp3s to playback on my handsets (this includes music on hold). I setup a couple of extensions, 901 and 902 to playback an mp3 I loaded on, and the sample moh that is included with Asterisk. When I attempt to call either extension I dont hear any sound, and the following displays on the console: -- Executing Answer(SIP/931-0efa, ) in new stack -- Executing Wait(SIP/931-0efa, 1) in new stack -- Executing MP3Player(SIP/931-0efa, /var/lib/asterisk/mohmp3/sample-hold .mp3) in new stack Jan 22 11:12:32 WARNING[442386]: rtp.c:375 ast_rtp_read: RTP Read error: Resourc e temporarily unavailable Jan 22 11:12:35 NOTICE[442386]: app_mp3.c:93 timed_read: Selected timed out/erro red out with 0 -- Executing Wait(SIP/931-0efa, 20) in new stack == Spawn extension (local, 902, 4) exited non-zero on 'SIP/931-0efa' The IP phones Im using are Cisco 7940 running G.729a. I have successfully licenced and registered 2x channels of g729 codec (running the new_codec_binary from ftp.digium.com) today, and have no problems checking my voicemail on Asterisk or dialing out through IAXtel or receiving calls. Even when I was running g711ulaw codec on the phones I had the same problem. Is there another dependency that is required for mp3playback in Linux? Is a soundcard required? My Linux box is running Slackware Linux 9.1. Any help to point me in the right direction to getting mp3playback and my music on hold working would be greatly appreciated. Thanks in advance, Chris Lee
[Asterisk-Users] mp3player not working
Hi, Im running the latest Asterisk (built last Saturday) and cant get mp3s to playback on my handsets (this includes music on hold). I setup a couple of extensions, 901 and 902 to playback an mp3 I loaded on, and the sample moh that is included with Asterisk. When I attempt to call either extension I dont hear any sound, and the following displays on the console: -- Executing Answer(SIP/931-0efa, ) in new stack -- Executing Wait(SIP/931-0efa, 1) in new stack -- Executing MP3Player(SIP/931-0efa, /var/lib/asterisk/mohmp3/sample-hold .mp3) in new stack Jan 22 11:12:32 WARNING[442386]: rtp.c:375 ast_rtp_read: RTP Read error: Resourc e temporarily unavailable Jan 22 11:12:35 NOTICE[442386]: app_mp3.c:93 timed_read: Selected timed out/erro red out with 0 -- Executing Wait(SIP/931-0efa, 20) in new stack == Spawn extension (local, 902, 4) exited non-zero on 'SIP/931-0efa' The IP phones Im using are Cisco 7940 running G.729a. I have successfully licenced and registered 2x channels of g729 codec (running the new_codec_binary from ftp.digium.com) today, and have no problems checking my voicemail on Asterisk or dialing out through IAXtel or receiving calls. Even when I was running g711ulaw codec on the phones I had the same problem. Is there another dependency that is required for mp3playback in Linux? Is a soundcard required? My Linux box is running Slackware Linux 9.1. Any help to point me in the right direction to getting mp3playback and my music on hold working would be greatly appreciated. Thanks in advance, Chris Lee
Re: [Asterisk-Users] MP3Player problem -repost
On 17 Nov 2003 13:39:20 +0100, Areski [EMAIL PROTECTED] wrote: I tried also to enter directly this instruction: mpg123 -w ki.wav http://digitaljukebox.com/Carta.mp3 And I get : HTTP request failed: HTML PUBLIC -//IETF//DTD HTML 2.0//EN The file exist, I get do a wget on it... Some ideas how to get it working ??? It looks like it's doing a redirect from digitaljukebox.com to www.digitaljukebox.com: [EMAIL PROTECTED]:~$ telnet digitaljukebox.com 80 Trying 216.98.141.3... Connected to 4h1413.aspadmin.net. Escape character is '^]'. HEAD /Carta.mp3 HTTP/1.0 Host: digitaljukebox.com HTTP/1.1 302 Found Date: Mon, 17 Nov 2003 22:48:25 GMT Server: Apache/1.3.20 Sun Cobalt (Unix) mod_ssl/2.8.4 OpenSSL/0.9.6b PHP/4.0.6 mod_auth_pam_external/0.1 FrontPage/4.0.4.3 mod_perl/1.25 Location: http://www.digitaljukebox.com/Carta.mp3 Connection: close Content-Type: text/html; charset=iso-8859-1 Connection closed by foreign host. Try using www.digitaljukebox.com instead. mpg123 has a simple mind and likes things to be simple HTTP-wise. -rt -- Ryan Tucker Network Engineer NetAccess, Inc. 1159 Pittsford-Victor Road Bldg. 5, Suite 140 Pittsford, New York 14534 585-419-8200 www.netacc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MP3Player problem
Hi all, Finally, I replaced the version of mpg123 by the last one and now it's working well to play local mp3 file... Well, now I m trying to stream a no-local file in my AGI script: EXEC MP3Player \http://digitaljukebox.com/Carta.mp3\; Is it possible to do ??? Thanks in advance, Aresk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MP3Player problem
Hi all, Finally, I replaced the version of mpg123 by the last one and now it's working well to play local mp3 file... Well, now I m trying to stream a no-local file in my AGI script: EXEC MP3Player \http://digitaljukebox.com/Carta.mp3\; Is it possible to do ??? Thanks in advance, Aresk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MP3Player problem
On Fri, 2003-11-07 at 20:27, Ernest W. Lessenger wrote: At 09:20 AM 11/7/2003, you wrote: I though to it also, but really I don't know how can I get the pid of a process ran by asterisk. I mean, the only think I do it's :print EXEC MP3Player \$key\\n; Then asterisk take the hand with mp3player applications that will launch mpg123, etc... You're right, I thought you were using EAGI... Well, I can use EAGI, but I still don't know how to get the pid in this script... zup =( That app seems to work fine for me. MPG123 does seem to be giving asterisk trouble, generally. I tried a few different changes, but nothing resulted in a fix. Is MPG123 maybe not responding to a SIGKILL properly? It doesn't seem, I use to kill the process manually after the test and it's working (kill -9 pid) When asterisk run mpg123, there is two process running (one is forked probably), it seem that one is well killed by asterisk and the other one is getting crazy when the other die, in fact it start to take a lot of cpu load... I m starting to think that the problem is coming from mpg123! Thx for u help, Aresk --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MP3Player problem
Hi ! Now I can hear nice mp3 through my phone... Great :P And many thanks for your posts. Now it's working fine... hmmm almost !!! In fact, I m using DialenMP3.agi. It's a real nice agi script... For those which would not know, you will find it here : http://asterisk.gnuinter.net/files/digium/asterisk-ng/agi/ What the agi script does, it's EXEC MP3Player \$key\\n; in which the key is the mp3 to play. Well the things weird, really weird, it's that the process, launched by the agi script, is never killed !!! Any ideas to fix that ?!? Thanks in advance, Areski On Thu, 2003-11-06 at 21:25, Rich Adamson wrote: Take a look at: http://www.voip-info.org/wiki-Asterisk+mpg123+redhat as there are steps for installation, and a sample CLI of a working MOH system that might be useful. I've never had both installed, but I heard that all life will cease to exist, or something else really bad... Can't remember exactly what it was ;) But all kidding around aside, I read in one of the archives that having both installed kept it from working right. So my suggestion is to uninstall 321. -Original Message- Well, thx for your answer... In fact, I have the both installed ! I don't think that it s a problem, isn't it ? Regards, Areski On Thu, 2003-11-06 at 17:25, David Gomillion wrote: Make sure you have mpg123 installed instead of mpg321... It's in the archives somewhere... that's what fixed my install. HTH, David Gomillion -Original Message- Hi all, Is there something wrong with MP3Player ??? I always get the message below when I try to play a MP3 : -- Executing MP3Player(SIP/phone1-83f9, /var/lib/asterisk/mohmp3/02) in new stack WARNING[79885]: File rtp.c, Line 374 (ast_rtp_read): RTP Read error: Resource temporarily unavailable NOTICE[79885]: File app_mp3.c, Line 93 (timed_read): Selected timed out/errored out with 0 I already saw some old posts about it but no solutions ! part of an old post : I found that executing the new mpg123 with: mpg123 sample-hold.mp3 = sometimes takes a couple of seconds to start playing. Every subsequent = command (exactly the same) starts playing immediately. Maybe this causes = the timeout in *? Can anyone give me a direction to solve this problem ? Thanks in advance, Areski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MP3Player problem
Hi ! Now I can hear nice mp3 through my phone... Great :P And many thanks for your posts. Now it's working fine... hmmm almost !!! In fact, I m using DialenMP3.agi. It's a real nice agi script... For those which would not know, you will find it here : http://asterisk.gnuinter.net/files/digium/asterisk-ng/agi/ What the agi script does, it's EXEC MP3Player \$key\\n; in which the key is the mp3 to play. Well the things weird, really weird, it's that the process, launched by the agi script, is never killed !!! Any ideas to fix that ?!? Thanks in advance, Areski On Thu, 2003-11-06 at 21:25, Rich Adamson wrote: Take a look at: http://www.voip-info.org/wiki-Asterisk+mpg123+redhat as there are steps for installation, and a sample CLI of a working MOH system that might be useful. I've never had both installed, but I heard that all life will cease to exist, or something else really bad... Can't remember exactly what it was ;) But all kidding around aside, I read in one of the archives that having both installed kept it from working right. So my suggestion is to uninstall 321. -Original Message- Well, thx for your answer... In fact, I have the both installed ! I don't think that it s a problem, isn't it ? Regards, Areski On Thu, 2003-11-06 at 17:25, David Gomillion wrote: Make sure you have mpg123 installed instead of mpg321... It's in the archives somewhere... that's what fixed my install. HTH, David Gomillion -Original Message- Hi all, Is there something wrong with MP3Player ??? I always get the message below when I try to play a MP3 : -- Executing MP3Player(SIP/phone1-83f9, /var/lib/asterisk/mohmp3/02) in new stack WARNING[79885]: File rtp.c, Line 374 (ast_rtp_read): RTP Read error: Resource temporarily unavailable NOTICE[79885]: File app_mp3.c, Line 93 (timed_read): Selected timed out/errored out with 0 I already saw some old posts about it but no solutions ! part of an old post : I found that executing the new mpg123 with: mpg123 sample-hold.mp3 = sometimes takes a couple of seconds to start playing. Every subsequent = command (exactly the same) starts playing immediately. Maybe this causes = the timeout in *? Can anyone give me a direction to solve this problem ? Thanks in advance, Areski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MP3Player problem
At 07:01 AM 11/7/2003, you wrote: Hi ! Now I can hear nice mp3 through my phone... Great :P And many thanks for your posts. Now it's working fine... hmmm almost !!! In fact, I m using DialenMP3.agi. It's a real nice agi script... For those which would not know, you will find it here : http://asterisk.gnuinter.net/files/digium/asterisk-ng/agi/ What the agi script does, it's EXEC MP3Player \$key\\n; in which the key is the mp3 to play. Well the things weird, really weird, it's that the process, launched by the agi script, is never killed !!! Any ideas to fix that ?!? You need to... 1) Set up a signal handler to handle the case when the mp3 player dies before you are ready (SIGCHILD) 2) Kill the mp3 player before you exit the AGI script (kill procid) I'll help you with this if you need it, it's really not all that hard. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MP3Player problem
I though to it also, but really I don't know how can I get the pid of a process ran by asterisk. I mean, the only think I do it's :print EXEC MP3Player \$key\\n; Then asterisk take the hand with mp3player applications that will launch mpg123, etc... How can I get this pid of the good mpg123 process ? (even if there is more than one mpg123 running...) If you can give me a hand or a direction, it would be really great, Thx, Areski On Fri, 2003-11-07 at 17:56, Ernest W. Lessenger wrote: At 07:01 AM 11/7/2003, you wrote: Hi ! Now I can hear nice mp3 through my phone... Great :P And many thanks for your posts. Now it's working fine... hmmm almost !!! In fact, I m using DialenMP3.agi. It's a real nice agi script... For those which would not know, you will find it here : http://asterisk.gnuinter.net/files/digium/asterisk-ng/agi/ What the agi script does, it's EXEC MP3Player \$key\\n; in which the key is the mp3 to play. Well the things weird, really weird, it's that the process, launched by the agi script, is never killed !!! Any ideas to fix that ?!? You need to... 1) Set up a signal handler to handle the case when the mp3 player dies before you are ready (SIGCHILD) 2) Kill the mp3 player before you exit the AGI script (kill procid) I'll help you with this if you need it, it's really not all that hard. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MP3Player problem
At 09:20 AM 11/7/2003, you wrote: I though to it also, but really I don't know how can I get the pid of a process ran by asterisk. I mean, the only think I do it's :print EXEC MP3Player \$key\\n; Then asterisk take the hand with mp3player applications that will launch mpg123, etc... You're right, I thought you were using EAGI... That app seems to work fine for me. MPG123 does seem to be giving asterisk trouble, generally. I tried a few different changes, but nothing resulted in a fix. Is MPG123 maybe not responding to a SIGKILL properly? --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MP3Player problem
Hi all, Is there something wrong with MP3Player ??? I always get the message below when I try to play a MP3 : -- Executing MP3Player(SIP/phone1-83f9, /var/lib/asterisk/mohmp3/02) in new stack WARNING[79885]: File rtp.c, Line 374 (ast_rtp_read): RTP Read error: Resource temporarily unavailable NOTICE[79885]: File app_mp3.c, Line 93 (timed_read): Selected timed out/errored out with 0 I already saw some old posts about it but no solutions ! part of an old post : I found that executing the new mpg123 with: mpg123 sample-hold.mp3 = sometimes takes a couple of seconds to start playing. Every subsequent = command (exactly the same) starts playing immediately. Maybe this causes = the timeout in *? Can anyone give me a direction to solve this problem ? Thanks in advance, Areski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MP3Player problem
Make sure you have mpg123 installed instead of mpg321... It's in the archives somewhere... that's what fixed my install. HTH, David Gomillion -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Areski Sent: Thursday, November 06, 2003 9:18 AM To: Asterisk-Users Mailing-list Subject: [Asterisk-Users] MP3Player problem Hi all, Is there something wrong with MP3Player ??? I always get the message below when I try to play a MP3 : -- Executing MP3Player(SIP/phone1-83f9, /var/lib/asterisk/mohmp3/02) in new stack WARNING[79885]: File rtp.c, Line 374 (ast_rtp_read): RTP Read error: Resource temporarily unavailable NOTICE[79885]: File app_mp3.c, Line 93 (timed_read): Selected timed out/errored out with 0 I already saw some old posts about it but no solutions ! part of an old post : I found that executing the new mpg123 with: mpg123 sample-hold.mp3 = sometimes takes a couple of seconds to start playing. Every subsequent = command (exactly the same) starts playing immediately. Maybe this causes = the timeout in *? Can anyone give me a direction to solve this problem ? Thanks in advance, Areski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MP3Player problem
Well, thx for your answer... In fact, I have the both installed ! I don't think that it s a problem, isn't it ? Regards, Areski On Thu, 2003-11-06 at 17:25, David Gomillion wrote: Make sure you have mpg123 installed instead of mpg321... It's in the archives somewhere... that's what fixed my install. HTH, David Gomillion -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Areski Sent: Thursday, November 06, 2003 9:18 AM To: Asterisk-Users Mailing-list Subject: [Asterisk-Users] MP3Player problem Hi all, Is there something wrong with MP3Player ??? I always get the message below when I try to play a MP3 : -- Executing MP3Player(SIP/phone1-83f9, /var/lib/asterisk/mohmp3/02) in new stack WARNING[79885]: File rtp.c, Line 374 (ast_rtp_read): RTP Read error: Resource temporarily unavailable NOTICE[79885]: File app_mp3.c, Line 93 (timed_read): Selected timed out/errored out with 0 I already saw some old posts about it but no solutions ! part of an old post : I found that executing the new mpg123 with: mpg123 sample-hold.mp3 = sometimes takes a couple of seconds to start playing. Every subsequent = command (exactly the same) starts playing immediately. Maybe this causes = the timeout in *? Can anyone give me a direction to solve this problem ? Thanks in advance, Areski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MP3Player problem
Take a look at: http://www.voip-info.org/wiki-Asterisk+mpg123+redhat as there are steps for installation, and a sample CLI of a working MOH system that might be useful. I've never had both installed, but I heard that all life will cease to exist, or something else really bad... Can't remember exactly what it was ;) But all kidding around aside, I read in one of the archives that having both installed kept it from working right. So my suggestion is to uninstall 321. -Original Message- Well, thx for your answer... In fact, I have the both installed ! I don't think that it s a problem, isn't it ? Regards, Areski On Thu, 2003-11-06 at 17:25, David Gomillion wrote: Make sure you have mpg123 installed instead of mpg321... It's in the archives somewhere... that's what fixed my install. HTH, David Gomillion -Original Message- Hi all, Is there something wrong with MP3Player ??? I always get the message below when I try to play a MP3 : -- Executing MP3Player(SIP/phone1-83f9, /var/lib/asterisk/mohmp3/02) in new stack WARNING[79885]: File rtp.c, Line 374 (ast_rtp_read): RTP Read error: Resource temporarily unavailable NOTICE[79885]: File app_mp3.c, Line 93 (timed_read): Selected timed out/errored out with 0 I already saw some old posts about it but no solutions ! part of an old post : I found that executing the new mpg123 with: mpg123 sample-hold.mp3 = sometimes takes a couple of seconds to start playing. Every subsequent = command (exactly the same) starts playing immediately. Maybe this causes = the timeout in *? Can anyone give me a direction to solve this problem ? Thanks in advance, Areski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MP3Player and Ringing (long)
[I'm reposting this to the asterisk-users list, since it seems to be a bit more active.] Hello, I started messing with Asterisk few days ago, so my overall knoledge about it is still fairy superficial. I think I found an issue with MP3Player; it can be reproducted with this extension: exten = 6001,1,Answer exten = 6001,2,Background(blahblah) exten = 6001,3,Ringing exten = 6001,4,Wait(2) exten = 6001,5,MP3Player(pippo.mp3) What I think it is happening is that Background plays its GSM file setting the write format to 2 (GSM), Ringing starts the generator setting the write format to 64 (Linear PCM) saving the old write format 2; MP3Player starts, set the write format to 64. When it calls ast_write to write the first stream, ast_release_generator is called and restores the write format to 2 (GSM); further writes produce codec errors since MP3Players writes Linear PCM frames. I extracted some debug log: Jun 5 01:55:33 DEBUG[1158913328]: File chan_sip.c, Line 1327 (sip_alloc): Allocating new SIP call for [EMAIL PROTECTED] Jun 5 01:55:33 DEBUG[1158913328]: File chan_sip.c, Line 3359 (check_user): Setting NAT on RTP to 0 Jun 5 01:55:33 DEBUG[1158913328]: File chan_sip.c, Line 2899 (build_route): build_route: Contact hop: 5010 sip:[EMAIL PROTECTED] Jun 5 01:55:33 DEBUG[1236360496]: File pbx.c, Line 1116 (pbx_extension_helper): Launching 'Answer' Jun 5 01:55:33 DEBUG[1236360496]: File chan_sip.c, Line 934 (sip_answer): sip_answer(SIP/5010-d3c4) Jun 5 01:55:33 DEBUG[1236360496]: File pbx.c, Line 1116 (pbx_extension_helper): Launching 'BackGround' Jun 5 01:55:33 DEBUG[1236360496]: File channel.c, Line 1381 (ast_set_write_format): Set channel SIP/5010-d3c4 to write format 2 Jun 5 01:55:33 DEBUG[1236360496]: File rtp.c, Line 838 (ast_rtp_write): Ooh, format changed from 0 to 4 Jun 5 01:55:36 DEBUG[1158913328]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 26024: Found Jun 5 01:55:36 DEBUG[1158913328]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 26024: Not Found Jun 5 01:55:36 DEBUG[1158913328]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 26024: Not Found Jun 5 01:55:37 DEBUG[1236360496]: File pbx.c, Line 1116 (pbx_extension_helper): Launching 'Ringing' Jun 5 01:55:37 DEBUG[1236360496]: File channel.c, Line 1163 (ast_indicate): Driver for channel 'SIP/5010-d3c4' does not support indication 3, emulating it Jun 5 01:55:37 DEBUG[1236360496]: File channel.c, Line 747 (ast_activate_generator): ast_activate_generator Jun 5 01:55:37 DEBUG[1236360496]: File channel.c, Line 1381 (ast_set_write_format): Set channel SIP/5010-d3c4 to write format 64 Jun 5 01:55:37 DEBUG[1236360496]: File pbx.c, Line 1116 (pbx_extension_helper): Launching 'Wait' Jun 5 01:55:37 DEBUG[1158913328]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 26024: Not Found Jun 5 01:55:38 DEBUG[1236360496]: File pbx.c, Line 1116 (pbx_extension_helper): Launching 'MP3Player' Jun 5 01:55:38 DEBUG[1236360496]: File channel.c, Line 1381 (ast_set_write_format): Set channel SIP/5010-d3c4 to write format 64 Jun 5 01:55:39 DEBUG[1236360496]: File channel.c, Line 1298 (ast_write): ast_write - ast_deactivate_generator Jun 5 01:55:39 DEBUG[1236360496]: File channel.c, Line 734 (ast_deactivate_generator): ast_deactivate_generator Jun 5 01:55:39 DEBUG[1236360496]: File indications.c, Line 62 (playtones_release): playtones_release Jun 5 01:55:39 DEBUG[1236360496]: File channel.c, Line 1381 (ast_set_write_format): Set channel SIP/5010-d3c4 to write format 2 After that, the console is filled with: WARNING[1236360496]: File codec_gsm.c, Line 136 (gsmtolin_framein): Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from mp3_exec (320) I attempted to dig into this thing, if I used another call to Background (even with another format) the problem didn't show up since ast_openstream takes care of calling ast_deactivate_generator, so I simply added: if (chan-generator) ast_deactivate_generator(chan); to app_mp3.c and the problem seems to go away. I think that the real question is if it is wise to rely on a side effect of ast_openstream in applications and on the other side if it is wise to delegate the to the application the call to ast_deactivate_generator. I have the feeling that there's a lack of precise semantic in the API to access channels and in my experience this often is the cause of subtle bugs. Of course, since I think I'm missing many aspects of Asterisk I may be completely wrong, in this case I'd appreciate very much some clarification about the interactions between applications, generator and streams with respect to the channel. Thanks in advance to everyone who took the time to read this :^) Bye! -- Daniele Orlandi ___ Asterisk-Users mailing list
[Asterisk-Users] MP3Player
Hi all, Finally I found why MP3Player was not working for me. In the CVS of two weeks ago the path to mpg123 was hardcoded to /usr/bin/mpg123. I installed the latest pre0.59s because previous releases were not working for me because of my fast Pentium IV 1,7Ghz processor. This release but probably previous releases also installs in /usr/local/bin/mpg123. What are the prerequisites for MP3's? The sample-hold.mp3 plays fine but my own mp3's segfault. Regards Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3Player
Hi all,
I found that executing the new mpg123 with: mpg123
sample-hold.mp3 sometimes takes a couple of seconds to start playing. Every
subsequent command (exactly the same) starts playing immediately. Maybe this
causes the timeout in *?
Regards Jan.
- Original Message -
From:
Jan Boon
To: [EMAIL PROTECTED]
Sent: Tuesday, June 03, 2003 12:25
PM
Subject: Re: [Asterisk-Users]
MP3Player
Hi all,
Unfortunately no luck:
-- Executing MP3Player("Modem[i4l]/ttyI0",
"/var/lib/asterisk/mohmp3/sample-hold.mp3") in new stack
NOTICE[131084]: File app_mp3.c, Line 80 (timed_read): Selected timed
out/errored out with 0
-- Timeout on Modem[i4l]/ttyI0
-- Executing MP3Player("SIP/1101-f902",
"/var/lib/asterisk/mohmp3/sample-hold.mp3") in new stack
NOTICE[172044]: File app_mp3.c, Line 80 (timed_read): Selected timed
-- Timeout on SIP/1101-f902
Is my processor Pentium IV 1,7Ghz 512Mb 266Mhz RAM too
fast for some timed functions like with mpg123? Are there compile options for
*?
Regards Jan.
- Original Message -
From:
Jan Boon
To: [EMAIL PROTECTED]
Sent: Tuesday, June 03, 2003 9:08
AM
Subject: Re: [Asterisk-Users]
MP3Player
Hi all,
There was something wrong. With mpg123
(pre0.59s) there is a new option make linux-pentium. This produces wav files
with the correct length and now I can hearthose in players. Next step:
Try out in *.
Regards Jan.
- Original Message -
From:
Jan Boon
To: [EMAIL PROTECTED]
Sent: Tuesday, June 03, 2003 8:19
AM
Subject: Re: [Asterisk-Users]
MP3Player
Hi all,
Maybe something is wrong with my mpg123
(0.59r). When I make a wav file of sample-hold.mp3 with mpg123 -w x.wav -r
8000 sample-hold.mp3 I see that the 28 seconds of the original clip has
been shortened to half of it (14 seconds). The sound recorder is playing
recorded wav files fine but I hear nothingof the wav file generated
by mpg123.
Hope to hear some clues.
Regards Jan.
- Original Message -
From:
Jan
Boon
To: [EMAIL PROTECTED]
Sent: Monday, June 02, 2003 8:11
PM
Subject: [Asterisk-Users]
MP3Player
Hi all,
Here another guy working with ztdummy and
having problems with music.
MP3Player does not work for me both from a
telephone entering through a passive ISDN-adapter as well as a
SIP-client in the LAN. Ztdummy works with conferences.
Here are my messages.
I have seen several discussion threads in
the mail-archive regarding this subject, but could not find an
answer.
Is this something I have to do without
without Zaptel devices?
-- Executing
BackGround("Modem[i4l]/ttyI0", "demo-instruct") in new
stack -- Playing
'demo-instruct' -- Executing
MP3Player("Modem[i4l]/ttyI0",
"/var/lib/asterisk/mohmp3/sample-hold.mp3") in new
stackNOTICE[73738]: File app_mp3.c, Line 80 (timed_read): Selected
timed out/errored out with 0 -- Timeout on
Modem[i4l]/ttyI0
-- Hungup
'Modem[i4l]/ttyI0' -- Registered SIP '' at
192.168.17.101 port 12308 expires 1200 --
Executing Ringing("SIP/1101-adfc", "") in new
stack -- Executing Answer("SIP/1101-adfc", "") in
new stack -- Executing MP3Player("SIP/1101-adfc",
"/var/lib/asterisk/mohmp3/sample-hold.mp3") in new
stackWARNING[90122]: File rtp.c, Line 288 (ast_rtp_read): RTP Read
error: Resource temporarily unavailableNOTICE[90122]: File
app_mp3.c, Line 80 (timed_read): Selected timed out/errored out with
0 -- Timeout on
SIP/1101-adfc -- Executing Hangup("SIP/1101-adfc",
"") in new stack
Regards Jan.
PS. How can I play a MP3-file through my
speakers with mpg123? I have Alsa and OSS-emulation. MPlayer and Xine do
this real
fine.
[Asterisk-Users] MP3Player
Hi all,
Here another guy working with ztdummy and having
problems with music.
MP3Player does not work for me both from a
telephone entering through a passive ISDN-adapter as well as a SIP-client in the
LAN. Ztdummy works with conferences.
Here are my messages.
I have seen several discussion threads in the
mail-archive regarding this subject, but could not find an answer.
Is this something I have to do without without
Zaptel devices?
-- Executing
BackGround("Modem[i4l]/ttyI0", "demo-instruct") in new
stack -- Playing 'demo-instruct' --
Executing MP3Player("Modem[i4l]/ttyI0",
"/var/lib/asterisk/mohmp3/sample-hold.mp3") in new stackNOTICE[73738]: File
app_mp3.c, Line 80 (timed_read): Selected timed out/errored out with
0 -- Timeout on Modem[i4l]/ttyI0
-- Hungup
'Modem[i4l]/ttyI0' -- Registered SIP '' at 192.168.17.101
port 12308 expires 1200 -- Executing
Ringing("SIP/1101-adfc", "") in new stack -- Executing
Answer("SIP/1101-adfc", "") in new stack -- Executing
MP3Player("SIP/1101-adfc", "/var/lib/asterisk/mohmp3/sample-hold.mp3") in new
stackWARNING[90122]: File rtp.c, Line 288 (ast_rtp_read): RTP Read error:
Resource temporarily unavailableNOTICE[90122]: File app_mp3.c, Line 80
(timed_read): Selected timed out/errored out with 0 --
Timeout on SIP/1101-adfc -- Executing
Hangup("SIP/1101-adfc", "") in new stack
Regards Jan.
PS. How can I play a MP3-file through my speakers
with mpg123? I have Alsa and OSS-emulation. MPlayer and Xine do this real
fine.
Re: [Asterisk-Users] MP3Player
Hi all,
Maybe something is wrong with my mpg123 (0.59r).
When I make a wav file of sample-hold.mp3 with mpg123 -w x.wav -r 8000
sample-hold.mp3 I see that the 28 seconds of the original clip has been
shortened to half of it (14 seconds). The sound recorder is playing recorded wav
files fine but I hear nothingof the wav file generated by
mpg123.
Hope to hear some clues.
Regards Jan.
- Original Message -
From:
Jan Boon
To: [EMAIL PROTECTED]
Sent: Monday, June 02, 2003 8:11 PM
Subject: [Asterisk-Users] MP3Player
Hi all,
Here another guy working with ztdummy and having
problems with music.
MP3Player does not work for me both from a
telephone entering through a passive ISDN-adapter as well as a SIP-client in
the LAN. Ztdummy works with conferences.
Here are my messages.
I have seen several discussion threads in the
mail-archive regarding this subject, but could not find an
answer.
Is this something I have to do without without
Zaptel devices?
-- Executing
BackGround("Modem[i4l]/ttyI0", "demo-instruct") in new
stack -- Playing 'demo-instruct'
-- Executing MP3Player("Modem[i4l]/ttyI0",
"/var/lib/asterisk/mohmp3/sample-hold.mp3") in new stackNOTICE[73738]:
File app_mp3.c, Line 80 (timed_read): Selected timed out/errored out with
0 -- Timeout on Modem[i4l]/ttyI0
-- Hungup
'Modem[i4l]/ttyI0' -- Registered SIP '' at
192.168.17.101 port 12308 expires 1200 -- Executing
Ringing("SIP/1101-adfc", "") in new stack -- Executing
Answer("SIP/1101-adfc", "") in new stack -- Executing
MP3Player("SIP/1101-adfc", "/var/lib/asterisk/mohmp3/sample-hold.mp3") in new
stackWARNING[90122]: File rtp.c, Line 288 (ast_rtp_read): RTP Read error:
Resource temporarily unavailableNOTICE[90122]: File app_mp3.c, Line 80
(timed_read): Selected timed out/errored out with 0 --
Timeout on SIP/1101-adfc -- Executing
Hangup("SIP/1101-adfc", "") in new stack
Regards Jan.
PS. How can I play a MP3-file through my speakers
with mpg123? I have Alsa and OSS-emulation. MPlayer and Xine do this real
fine.
Re: [Asterisk-Users] MP3Player
Hi all,
There was something wrong. With mpg123 (pre0.59s)
there is a new option make linux-pentium. This produces wav files with the
correct length and now I can hearthose in players. Next step: Try out in
*.
Regards Jan.
- Original Message -
From:
Jan Boon
To: [EMAIL PROTECTED]
Sent: Tuesday, June 03, 2003 8:19
AM
Subject: Re: [Asterisk-Users]
MP3Player
Hi all,
Maybe something is wrong with my mpg123 (0.59r).
When I make a wav file of sample-hold.mp3 with mpg123 -w x.wav -r 8000
sample-hold.mp3 I see that the 28 seconds of the original clip has been
shortened to half of it (14 seconds). The sound recorder is playing recorded
wav files fine but I hear nothingof the wav file generated by
mpg123.
Hope to hear some clues.
Regards Jan.
- Original Message -
From:
Jan Boon
To: [EMAIL PROTECTED]
Sent: Monday, June 02, 2003 8:11
PM
Subject: [Asterisk-Users]
MP3Player
Hi all,
Here another guy working with ztdummy and
having problems with music.
MP3Player does not work for me both from a
telephone entering through a passive ISDN-adapter as well as a SIP-client in
the LAN. Ztdummy works with conferences.
Here are my messages.
I have seen several discussion threads in the
mail-archive regarding this subject, but could not find an
answer.
Is this something I have to do without without
Zaptel devices?
-- Executing
BackGround("Modem[i4l]/ttyI0", "demo-instruct") in new
stack -- Playing 'demo-instruct'
-- Executing MP3Player("Modem[i4l]/ttyI0",
"/var/lib/asterisk/mohmp3/sample-hold.mp3") in new stackNOTICE[73738]:
File app_mp3.c, Line 80 (timed_read): Selected timed out/errored out with
0 -- Timeout on Modem[i4l]/ttyI0
-- Hungup
'Modem[i4l]/ttyI0' -- Registered SIP '' at
192.168.17.101 port 12308 expires 1200 -- Executing
Ringing("SIP/1101-adfc", "") in new stack -- Executing
Answer("SIP/1101-adfc", "") in new stack -- Executing
MP3Player("SIP/1101-adfc", "/var/lib/asterisk/mohmp3/sample-hold.mp3") in
new stackWARNING[90122]: File rtp.c, Line 288 (ast_rtp_read): RTP Read
error: Resource temporarily unavailableNOTICE[90122]: File app_mp3.c,
Line 80 (timed_read): Selected timed out/errored out with
0 -- Timeout on SIP/1101-adfc --
Executing Hangup("SIP/1101-adfc", "") in new stack
Regards Jan.
PS. How can I play a MP3-file through my
speakers with mpg123? I have Alsa and OSS-emulation. MPlayer and Xine do
this real fine.
Re: [Asterisk-Users] MP3Player
Hi all,
Unfortunately no luck:
-- Executing MP3Player("Modem[i4l]/ttyI0",
"/var/lib/asterisk/mohmp3/sample-hold.mp3") in new stack
NOTICE[131084]: File app_mp3.c, Line 80 (timed_read): Selected timed
out/errored out with 0
-- Timeout on Modem[i4l]/ttyI0
-- Executing MP3Player("SIP/1101-f902",
"/var/lib/asterisk/mohmp3/sample-hold.mp3") in new stack
NOTICE[172044]: File app_mp3.c, Line 80 (timed_read): Selected timed
-- Timeout on SIP/1101-f902
Is my processor Pentium IV 1,7Ghz 512Mb 266Mhz RAM too fast
for some timed functions like with mpg123? Are there compile options for
*?
Regards Jan.
- Original Message -
From:
Jan Boon
To: [EMAIL PROTECTED]
Sent: Tuesday, June 03, 2003 9:08
AM
Subject: Re: [Asterisk-Users]
MP3Player
Hi all,
There was something wrong. With mpg123 (pre0.59s)
there is a new option make linux-pentium. This produces wav files with the
correct length and now I can hearthose in players. Next step: Try out in
*.
Regards Jan.
- Original Message -
From:
Jan Boon
To: [EMAIL PROTECTED]
Sent: Tuesday, June 03, 2003 8:19
AM
Subject: Re: [Asterisk-Users]
MP3Player
Hi all,
Maybe something is wrong with my mpg123
(0.59r). When I make a wav file of sample-hold.mp3 with mpg123 -w x.wav -r
8000 sample-hold.mp3 I see that the 28 seconds of the original clip has been
shortened to half of it (14 seconds). The sound recorder is playing recorded
wav files fine but I hear nothingof the wav file generated by
mpg123.
Hope to hear some clues.
Regards Jan.
- Original Message -
From:
Jan Boon
To: [EMAIL PROTECTED]
Sent: Monday, June 02, 2003 8:11
PM
Subject: [Asterisk-Users]
MP3Player
Hi all,
Here another guy working with ztdummy and
having problems with music.
MP3Player does not work for me both from a
telephone entering through a passive ISDN-adapter as well as a SIP-client
in the LAN. Ztdummy works with conferences.
Here are my messages.
I have seen several discussion threads in the
mail-archive regarding this subject, but could not find an
answer.
Is this something I have to do without
without Zaptel devices?
-- Executing
BackGround("Modem[i4l]/ttyI0", "demo-instruct") in new
stack -- Playing
'demo-instruct' -- Executing
MP3Player("Modem[i4l]/ttyI0", "/var/lib/asterisk/mohmp3/sample-hold.mp3")
in new stackNOTICE[73738]: File app_mp3.c, Line 80 (timed_read):
Selected timed out/errored out with 0 -- Timeout on
Modem[i4l]/ttyI0
-- Hungup
'Modem[i4l]/ttyI0' -- Registered SIP '' at
192.168.17.101 port 12308 expires 1200 -- Executing
Ringing("SIP/1101-adfc", "") in new stack --
Executing Answer("SIP/1101-adfc", "") in new stack
-- Executing MP3Player("SIP/1101-adfc",
"/var/lib/asterisk/mohmp3/sample-hold.mp3") in new
stackWARNING[90122]: File rtp.c, Line 288 (ast_rtp_read): RTP Read
error: Resource temporarily unavailableNOTICE[90122]: File app_mp3.c,
Line 80 (timed_read): Selected timed out/errored out with
0 -- Timeout on SIP/1101-adfc
-- Executing Hangup("SIP/1101-adfc", "") in new stack
Regards Jan.
PS. How can I play a MP3-file through my
speakers with mpg123? I have Alsa and OSS-emulation. MPlayer and Xine do
this real
fine.
[Asterisk-Users] MP3player problem
Hey,
I've installed 0.59r mpg123 on a redhatbox. I
set the extension up for the mp3player. I called and it was playing the file
back,but it was full of drops. like sound - silence - sound continued. I thought
let's try with the developer version of mpg123 cos other extensions like echo
test was working fine. so I've installed the 0.59s. Since then my * dies with
this message:
DEBUG[5126]: File chan_sip.c, Line 2773
(check_user): Setting NAT on RTP to 0 -- Executing
Answer("SIP/levisnom-7efd", "") in new stack -- Executing
MP3Player("SIP/levisnom-7efd", "/asterisk/c.mp3") in new
stackKilled
Any suggestion, how to solve this problem? Latest
cvs of course.
THX
__Levente
TamsICQ#:13692773
Current ICQ status:
+ More ways to contact me
__
online?icq=13692773img=21
Description: Binary data
smime.p7s
Description: S/MIME cryptographic signature
Re: [Asterisk-Users] MP3player problem
Can you use Playback instead ? Playback doesn't use mpg123. regards Martin On Thu, 3 Apr 2003, Tamas Levente wrote: Hey, I've installed 0.59r mpg123 on a redhat box. I set the extension up for the mp3player. I called and it was playing the file back,but it was full of drops. like sound - silence - sound continued. I thought let's try with the developer version of mpg123 cos other extensions like echo test was working fine. so I've installed the 0.59s. Since then my * dies with this message: DEBUG[5126]: File chan_sip.c, Line 2773 (check_user): Setting NAT on RTP to 0 -- Executing Answer(SIP/levisnom-7efd, ) in new stack -- Executing MP3Player(SIP/levisnom-7efd, /asterisk/c.mp3) in new stack Killed Any suggestion, how to solve this problem? Latest cvs of course. THX __ Levente Tamás ICQ#: 13692773 Current ICQ status: + More ways to contact me __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3player problem
And does playback support mp3? - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, April 03, 2003 5:58 PM Subject: Re: [Asterisk-Users] MP3player problem Can you use Playback instead ? Playback doesn't use mpg123. regards Martin On Thu, 3 Apr 2003, Tamas Levente wrote: Hey, I've installed 0.59r mpg123 on a redhat box. I set the extension up for the mp3player. I called and it was playing the file back,but it was full of drops. like sound - silence - sound continued. I thought let's try with the developer version of mpg123 cos other extensions like echo test was working fine. so I've installed the 0.59s. Since then my * dies with this message: DEBUG[5126]: File chan_sip.c, Line 2773 (check_user): Setting NAT on RTP to 0 -- Executing Answer(SIP/levisnom-7efd, ) in new stack -- Executing MP3Player(SIP/levisnom-7efd, /asterisk/c.mp3) in new stack Killed Any suggestion, how to solve this problem? Latest cvs of course. THX __ Levente Tamás ICQ#: 13692773 Current ICQ status: + More ways to contact me __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3player problem
what does playback use ? On Thursday 03 Apr 2003 16:18, Martin Pycko shaped the electrons to say: Can you use Playback instead ? Playback doesn't use mpg123. regards Martin On Thu, 3 Apr 2003, Tamas Levente wrote: Hey, I've installed 0.59r mpg123 on a redhat box. I set the extension up for the mp3player. I called and it was playing the file back,but it was full of drops. like sound - silence - sound continued. I thought let's try with the developer version of mpg123 cos other extensions like echo test was working fine. so I've installed the 0.59s. Since then my * dies with this message: DEBUG[5126]: File chan_sip.c, Line 2773 (check_user): Setting NAT on RTP to 0 -- Executing Answer(SIP/levisnom-7efd, ) in new stack -- Executing MP3Player(SIP/levisnom-7efd, /asterisk/c.mp3) in new stack Killed Any suggestion, how to solve this problem? Latest cvs of course. THX __ Levente Tamás ICQ#: 13692773 Current ICQ status: + More ways to contact me __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this correspondence in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or rely on any part of this correspondence if you are not the intended recipient. Any opinions expressed in this message are those of the individual sender. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3player problem
I woudln't write that if it wouldn't support mp3. On Thu, 3 Apr 2003, Tamas Levente wrote: And does playback support mp3? - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, April 03, 2003 5:58 PM Subject: Re: [Asterisk-Users] MP3player problem Can you use Playback instead ? Playback doesn't use mpg123. regards Martin On Thu, 3 Apr 2003, Tamas Levente wrote: Hey, I've installed 0.59r mpg123 on a redhat box. I set the extension up for the mp3player. I called and it was playing the file back,but it was full of drops. like sound - silence - sound continued. I thought let's try with the developer version of mpg123 cos other extensions like echo test was working fine. so I've installed the 0.59s. Since then my * dies with this message: DEBUG[5126]: File chan_sip.c, Line 2773 (check_user): Setting NAT on RTP to 0 -- Executing Answer(SIP/levisnom-7efd, ) in new stack -- Executing MP3Player(SIP/levisnom-7efd, /asterisk/c.mp3) in new stack Killed Any suggestion, how to solve this problem? Latest cvs of course. THX __ Levente Tamás ICQ#: 13692773 Current ICQ status: + More ways to contact me __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3player problem
This is why I asked. (the file is there) DEBUG[5126]: File chan_sip.c, Line 2773 (check_user): Setting NAT on RTP to 0 -- Executing Answer(SIP/levisnom-c18d, ) in new stack -- Executing Playback(SIP/levisnom-c18d, /asterisk/c.mp3) in new stack WARNING[16400]: File file.c, Line 410 (ast_openstream): File /asterisk/c.mp3 does not exist in any format WARNING[16400]: File file.c, Line 553 (ast_streamfile): Unable to open /asterisk/c.mp3 (format 4): No such file or directory WARNING[16400]: File app_playback.c, Line 83 (playback_exec): ast_streamfile failed on SIP/levisnom-c18d for /asterisk/c.mp3 DEBUG[5126]: File chan_sip.c, Line 460 (__sip_ack): Stopping retransmission on '3e8ca08db129-8h0t9jtcea0u@(null)' of Response 2: Found DEBUG[5126]: File chan_sip.c, Line 721 (__sip_destroy): Detaching from SIP/levisnom-c18d - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, April 03, 2003 7:44 PM Subject: Re: [Asterisk-Users] MP3player problem I woudln't write that if it wouldn't support mp3. On Thu, 3 Apr 2003, Tamas Levente wrote: And does playback support mp3? - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, April 03, 2003 5:58 PM Subject: Re: [Asterisk-Users] MP3player problem Can you use Playback instead ? Playback doesn't use mpg123. regards Martin On Thu, 3 Apr 2003, Tamas Levente wrote: Hey, I've installed 0.59r mpg123 on a redhat box. I set the extension up for the mp3player. I called and it was playing the file back,but it was full of drops. like sound - silence - sound continued. I thought let's try with the developer version of mpg123 cos other extensions like echo test was working fine. so I've installed the 0.59s. Since then my * dies with this message: DEBUG[5126]: File chan_sip.c, Line 2773 (check_user): Setting NAT on RTP to 0 -- Executing Answer(SIP/levisnom-7efd, ) in new stack -- Executing MP3Player(SIP/levisnom-7efd, /asterisk/c.mp3) in new stack Killed Any suggestion, how to solve this problem? Latest cvs of course. THX __ Levente Tamás ICQ#: 13692773 Current ICQ status: + More ways to contact me __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3player problem
If you use playback you don't specify the file ending. Since I only use the mp3 player for streaming content, I'm not sure if it holds true there also. On Thu, 2003-04-03 at 15:01, Tamas Levente wrote: This is why I asked. (the file is there) DEBUG[5126]: File chan_sip.c, Line 2773 (check_user): Setting NAT on RTP to 0 -- Executing Answer(SIP/levisnom-c18d, ) in new stack -- Executing Playback(SIP/levisnom-c18d, /asterisk/c.mp3) in new stack WARNING[16400]: File file.c, Line 410 (ast_openstream): File /asterisk/c.mp3 does not exist in any format WARNING[16400]: File file.c, Line 553 (ast_streamfile): Unable to open /asterisk/c.mp3 (format 4): No such file or directory WARNING[16400]: File app_playback.c, Line 83 (playback_exec): ast_streamfile failed on SIP/levisnom-c18d for /asterisk/c.mp3 DEBUG[5126]: File chan_sip.c, Line 460 (__sip_ack): Stopping retransmission on '3e8ca08db129-8h0t9jtcea0u@(null)' of Response 2: Found DEBUG[5126]: File chan_sip.c, Line 721 (__sip_destroy): Detaching from SIP/levisnom-c18d - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, April 03, 2003 7:44 PM Subject: Re: [Asterisk-Users] MP3player problem I woudln't write that if it wouldn't support mp3. On Thu, 3 Apr 2003, Tamas Levente wrote: And does playback support mp3? - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, April 03, 2003 5:58 PM Subject: Re: [Asterisk-Users] MP3player problem Can you use Playback instead ? Playback doesn't use mpg123. regards Martin On Thu, 3 Apr 2003, Tamas Levente wrote: Hey, I've installed 0.59r mpg123 on a redhat box. I set the extension up for the mp3player. I called and it was playing the file back,but it was full of drops. like sound - silence - sound continued. I thought let's try with the developer version of mpg123 cos other extensions like echo test was working fine. so I've installed the 0.59s. Since then my * dies with this message: DEBUG[5126]: File chan_sip.c, Line 2773 (check_user): Setting NAT on RTP to 0 -- Executing Answer(SIP/levisnom-7efd, ) in new stack -- Executing MP3Player(SIP/levisnom-7efd, /asterisk/c.mp3) in new stack Killed Any suggestion, how to solve this problem? Latest cvs of course. THX __ Levente Tamás ICQ#: 13692773 Current ICQ status: + More ways to contact me __ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3player problem
Playback does not take an extension of the file. It looks for the best file format to use. Try using it without the .mp3 extension on there. James On Thu, 3 Apr 2003, Tamas Levente wrote: This is why I asked. (the file is there) DEBUG[5126]: File chan_sip.c, Line 2773 (check_user): Setting NAT on RTP to 0 -- Executing Answer(SIP/levisnom-c18d, ) in new stack -- Executing Playback(SIP/levisnom-c18d, /asterisk/c.mp3) in new stack WARNING[16400]: File file.c, Line 410 (ast_openstream): File /asterisk/c.mp3 does not exist in any format WARNING[16400]: File file.c, Line 553 (ast_streamfile): Unable to open /asterisk/c.mp3 (format 4): No such file or directory WARNING[16400]: File app_playback.c, Line 83 (playback_exec): ast_streamfile failed on SIP/levisnom-c18d for /asterisk/c.mp3 DEBUG[5126]: File chan_sip.c, Line 460 (__sip_ack): Stopping retransmission on '3e8ca08db129-8h0t9jtcea0u@(null)' of Response 2: Found DEBUG[5126]: File chan_sip.c, Line 721 (__sip_destroy): Detaching from SIP/levisnom-c18d ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
