search for: 49443303

...flexibel rate not heavily tested! -- Started music on hold, class 'default', on Zap/4-1 Junk at the beginning 52494646 Skipped RIFF header! Warning, flexibel rate not heavily tested! -- Stopped music on hold on Zap/4-1 Warning, flexibel rate not heavily tested! Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested!

...exactly thus it reports this error, could help me? -- Executing WaitMusicOnHold("SIP/3205-d9ef", "30") in new stack -- Started music on hold, class 'default', on SIP/3205-d9ef Jun 23 02:14:21 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 Jun 23 02:14:21 WARNING[24960]: common.c:134 decode_header: Layer 1 not supported! Jun 23 02:14:21 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame e4e45a8a Jun 23 02:14:21 WARNING[24960]: layer3.c:966 III_dequantize_sample: mpg123: Can't rewind stream by 1758 bits! Jun...

...can't seem to get it started. This is a proof-of-concept installation, and currently does not have any T1/E1 or FXO/FXS cards in it. I just want to use it as an internal SIP server for now. However, when I try to start Asterisk, it dies with the following messages: Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe I have googled for info on these errors - some have reported getting them while _exiting_ Asterisk, but none have claimed to see them while _starting_ Asterisk. Any ideas? Thanks a ton. --sk.

...May 17 12:11:25 WARNING[13298]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 May 17 12:11:25 WARNING[13298]: loader.c:554 load_modules: Loading module chan_zap.so failed! sato:/etc/asterisk# Ouch ... error while writing audio data: : Broken pipe Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! Any help is greatly appreciated. Thanks, Jon Scottorn -------------- next part -------------- An HTML attachment was scrubbe...

...n state 1 Urgent handler Urgent handler -- Accepting overlap voice call from '4123655105' to '555115' on channel 0/26, span 4 -- Starting simple switch on 'Zap/119-1' Urgent handler Urgent handler Ouch ... error while writing audio data: : Broken pipe Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! Segmentation fault Are there any changes to the config neccessary? Are there known issues with this Asterisk version? Can I temporarily disable the TE411 specific stuff to get asterisk running again? Platform: HP DL380G3 with 2x2.4Ghz, 1GB RAM, 1xTE411P...

I have been attempting to download, compile and configure spandsp to function with * without much luck. I am guessing that some assumptions were made regarding the users knowledge level of Linux. Sadly, I must not live up to those assumptions. My problem begins when after compiling spandsp I look for the app_rxfax.c, app_txfax.c, app_dtmftotext.c and makefile.patch files to place in the

...p.so: load_module failed, returning -1 > == Unregistered channel type 'Tor' > == Unregistered channel type 'Zap' > -- Unregistered channel 1 > Apr 19 17:27:34 WARNING[16384]: loader.c:408 load_modules: Loading module chan_zap.so failed! > Junk at the beginning 49443303 > Can anyone out there using zaphfc, help me on this? Thanks in advance, --- Paulo Loureiro.

...E AT YOUR OWN RISK! Title : 10 - Track 10 Artist: <Unknown> Album : PROMO Year : Comment: Genre : Club Directory: /var/lib/asterisk/mohmp3/ Playing MPEG stream from TristeAlegriaPromo.mp3 ... Junk at the beginning 49443303 MPEG 1.0 layer III, 128 kbit/s, 44100 Hz joint-stereo [0:02] Decoding of TristeAlegriaPromo.mp3 finished. [root@copc ~]# ***************************************** Extensions.conf ***************************************** exten => 100,1,Answer exten => 100,2,Playback(tones-that-follow-are-f...

...======================= [root@smith root]# asterisk -c Asterisk CVS-v1-0-12/22/04-17:53:38, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster@digium.com> ========================================================================= [ Booting.................Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! .......................................................................[root@smith root]# Ouch ... error while writing audio data: : Broken pipe [root@smith root]# -------------- next part -------------- An HTML attachment was scrubbed... URL: http://li...

...(BTW, I have already secured proper license to play MOH to callers.) I used MS Media Player version 11 and rip it at 128kbps (smallest) but whenever I listen to MOH, I saw the following message on the Asterisk console. WARNING[20829]: mp3/interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 I tried it with different bit rate (320 kbps) and the same error message appeared. I used the following musiconhold.conf [classical] mode=files directory=/var/lib/asterisk/moh/classical random=yes Are there any Asterisk+Audio expert that can offer me some advice?

...E AT YOUR OWN RISK! Title : 10 - Track 10 Artist: <Unknown> Album : PROMO Year : Comment: Genre : Club Directory: /var/lib/asterisk/mohmp3/ Playing MPEG stream from TristeAlegriaPromo.mp3 ... Junk at the beginning 49443303 MPEG 1.0 layer III, 128 kbit/s, 44100 Hz joint-stereo [0:02] Decoding of TristeAlegriaPromo.mp3 finished. [root@copc ~]# ***************************************** Extensions.conf ***************************************** exten => 100,1,Answer exten => 100,2,Playback(tones-that-follow-are-f...

...musiconhold.so] => (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' Junk at the beginning 49443302 Warning, flexibel rate not heavily tested! Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! [res_adsi.so] => (ADSI Resource) [res_features.so] => (Call Parking Resource) == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls [res_crypto.so] => (Cryptograph...

...musiconhold.so] => (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' Junk at the beginning 49443302 Warning, flexibel rate not heavily tested! Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! [res_adsi.so] => (ADSI Resource) [res_features.so] => (Call Parking Resource) == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls [res_crypto.so] => (Cryptograph...

Folks, I've just successfully set up Asterisk (as part of the Asterisk Management Portal installation). When I say "successfully", I mean that I have gone through all the steps detailed for the installation of AMP and not hit any snags there. I can connect to my asterisk server via ssh and can also connect via Http to the portal to change settings in AMP. Now I'm trying to

I'm trying to create an extension that will connect caller to asterisk sound card. I've followed the example at http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+console. With no luck. What I get is: Jul 13 09:56:45 VERBOSE[1315]: -- Executing Dial("SIP/300-3bd6", "CONSOLE/dsp") in new stack Jul 13 09:56:45 WARNING[1315]: No channel type registered for