search for: app_mp3

Hi there, app_mp3 still does not work with the latest bri-stuff patch and the zaphfc driver. Here in my place it only works with the patch attached. For me it seems the bri-stuff worsens the asterisk timing... has anybody else made experiences with it? Deti -------------- next part -------------- Index: app_mp3....

...t 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 stack NOTICE[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 --...

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/maxhome@maxhome/3", "") in new stack -- Executing MP3Player...

...lt; -lrt endif [...] i get following error. Mar 21 19:08:22 WARNING[26686]: pbx.c:1720 pbx_extension_helper: No application 'mp4save' for extension... it seems that after the recompilation of asterisk no app_mp4.o/app_mp4.so is created in ../asterisk/apps/. asterisk# ls apps/app_mp* apps/app_mp3.c apps/app_mp3.o apps/app_mp3.so apps/app_mp4.c has anyone an idea... thx in advabce... ____________________________________________________________________________________ Sucker-punch spam with award-winning protection. Try the free Yahoo! Mail Beta. http://advision.webevents.yahoo.co...

There is a patch to mplayer that allows it to suppress stdout messages and instead output pcm data to stdout. I managed to get it working with app_mp3.c and seems like it is working fine. All that was needed was a change in the execl line and a slight increase in timeout value. I have only done limited testing. If mplayer is able to replace mpg123 without issues, this opens up a whole lot of media that can be piped through Asterisk. I want t...

...-- Executing MP3Player("SIP/27870-ba4f", "http://stream.lrz-muenchen.de:31337/m945-lq.mp3") in new stack Aug 2 18:16:16 DEBUG[262161]: chan_sip.c:817 __sip_ack: Stopping retransmission on '7e3ee6f44f3fabf0@*' of Response 27022: Found Aug 2 18:16:19 NOTICE[475156]: app_mp3.c:91 timed_read: Selected timed out/errored out with 0 Aug 2 18:16:19 DEBUG[475156]: app_mp3.c:175 mp3_exec: No more mp3 Playing back files from a directory works fine. Does anyone have any clues? My primary (and only) timing source is a pci-hfc card using the zaphfc driver and configured lik...

...l: cap_set_proc [May 12 21:36:55] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'app_ices.so': /usr/lib/asterisk/modules/app_ices.so: undefined symbol: cap_set_proc [May 12 21:36:55] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'app_mp3.so': /usr/lib/asterisk/modules/app_mp3.so: undefined symbol: cap_set_proc [May 12 21:36:55] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'app_nbscat.so': /usr/lib/asterisk/modules/app_nbscat.so: undefined symbol: cap_set_proc [May 12 21:36:55] WARNING[15344]...

...quot;, "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 I'm using are Cisco 7940 running G.729a. I have successfully l...

...on-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found WARNING[1142127920]: File chan_oss.c, Line 232 (sound_thread): Read error on sou nd device: Resource temporarily unavailable <snip> [app_mp3.so] => (Silly MP3 Application) == Registered application 'MP3Player' <snip> I can't seem to find any other reason for MOH to not work. If someone has MOH working, would it be possible for you to send me a "sip debug" listing of what's going on so I can compare...

...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://localh...

...great. Here's what I see in the console: -- Executing MP3Player("IAX2/income-in-01@IP", "/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 MPG...

Here's a question maybe someone can help me with: My extension looks like this: exten => 1006,1,MP3Player,http://audio-mp3.ibiblio.org:8000/wcpe.mp3 When I try this extension, the following output appears in the CLI: Nov 13 12:47:51 NOTICE[8422]: app_mp3.c:111 timed_read: Poll timed out/errored out with 0 I should mention that mpg123 is installed, however the server we are using for this project doesn't have an audio card. Is this a problem? Doesn't seem to be so far (everything else works great.) Cheers! Phil Jackson -- Phil Jackson CTO...

...|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

I have searching the forums here on how to get Music On Hold working and I have been able to get * to accept a command for MusicOnHold and for Meetme after loading the ztdummy module. I used the default config for /etc/zaptel.conf since I saw no guidance on this. My problem now is that when I activate MusicOnHold, the sample music file sounds very slow and distorted. My best guess is that it is

...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...

...oload => app_getcpeid.so load => app_groupcount.so load => app_hasnewvoicemail.so noload => app_ices.so noload => app_image.so noload => app_intercom.so load => app_lookupblacklist.so load => app_lookupcidname.so load => app_macro.so load => app_milliwatt.so load => app_mp3.so noload => app_nbscat.so load => app_parkandannounce.so load => app_playback.so load => app_privacy.so load => app_qcall.so load => app_queue.so load => app_random.so load => app_read.so load => app_record.so load => app_sayunixtime.so load => app_senddtmf.so load...

...ubstring digits in a 0 chan_skinny.so Skinny Client Control Protocol (Skinny) 0 format_sln.so Raw Signed Linear Audio support (SLN) 0 app_zapateller.so Block Telemarketers with Special Informa 0 app_queue.so True Call Queueing 0 app_mp3.so Silly MP3 Application 0 app_lookupcidname.so Look up CallerID Name from local databas 0 format_vox.so Dialogic VOX (ADPCM) File Format 0 chan_oss.so OSS Console Channel Driver 0 app_talkdetect.so Playb...

I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc. I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs pwlib openh323 gnugk for h.323 gatekeeper

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...from /usr/lib/asterisk/modules/app_voicemail.so...done. Loaded symbols for /usr/lib/asterisk/modules/app_voicemail.so Reading symbols from /usr/lib/asterisk/modules/app_directory.so...done. Loaded symbols for /usr/lib/asterisk/modules/app_directory.so Reading symbols from /usr/lib/asterisk/modules/app_mp3.so...done. Loaded symbols for /usr/lib/asterisk/modules/app_mp3.so Reading symbols from /usr/lib/asterisk/modules/app_system.so...done. Loaded symbols for /usr/lib/asterisk/modules/app_system.so Reading symbols from /usr/lib/asterisk/modules/app_echo.so...done. Loaded symbols for /usr/lib/asterisk/...