similar to: Junk at the beginning, Warning, flexibel rate not heavily tested!

Hello, I am currently using MusicOnHold(mpg123), and it works just fine, but every once in a while I will get a flurry of warnings in the CLI like those below and Asterisk will freeze completely, and the only way to come out of it is with a kill -9 . Is mpg123 causing my problem? Is there a specific format of MP3 that should be used/avoided to not have errors like these? Any help would be greatly

Hello. Is this normal. Or does it mean there is a problem ? ------------------------- stop now Beginning asterisk shutdown.... Executing last minute cleanups == Destroying any remaining musiconhold processes Yuck! Error in buffer handling...: Connection reset by peer Yuck! Error in buffer handling...: Broken pipe Yuck! Error in buffer handling...: Broken pipe Asterisk cleanly ending (0).

Hi. I am having the following warning when using music on hold. It works from X-Lite to Grandstream. I get a lot of errors and warnings. 1.Warning, flexibel rate not heavily tested! 2. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread: Request to schedule in the past?!?! Thanks for any help. Full Output below: Jan 30 10:24:55 WARNING[1133718080]: chan_sip.c:486

I'm an asterisk newbie, so initially tried to install it on a Mandrake 10.1 (see config below) and with a bit of messing about using sample config, have been able to make the test call to device 1000, and also through to the IAX test number at Digium. However, operation is extremely flaky - I cannot reliably startup and use the system on a regular basis. I have several problems listed below

Hi ! :) Does someone have an idea of the reason why my MusicOnHold (with clean mp3 downloaded from http://aussievoip.com.au/wiki-MOH) is always having interruptions and micro-cuts ? No high load of the system, no swap, no hard disk r/w, mpg123 seems running well... nothing ! Except a little message at startup : "Warning, flexibel rate not heavily tested!" I'm getting

Hello, This is debug output I get: Jul 28 15:05:49 WARNING[8249]: chan_oss.c:239 sound_thread: Read error on sound device: Resource temporarily unavailable [chan_zap.so] => (Zapata Telephony w/PRI) Jul 28 15:05:49 WARNING[8249]: chan_zap.c:924 zt_open: Unable to specify channel 1: No such device or address Jul 28 15:05:49 ERROR[8249]: chan_zap.c:6460 mkintf: Unable to open

Hello, I've done a very straightforward install of Asterisk, and 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

Hi List, I have a test asterisk box with a TDM400P with 4 FXO modules plugged in. Yesterday I could use the box without any issues - no problems. This morning, the sound on the box was absolutely horrible. After some fiddling about, I have rebooted the box, and now asterisk refuses to start! Here's the message I get: Jun 9 10:45:53 WARNING[3297]: chan_zap.c:769 zt_open: Unable to

Hi all, I replaced a TE410P Rev C 1st Generation Firmware with a TE411P without any cfg changes (zaptel/zapata). As a result Asterisk crashes on outbound from PRI4 going to PRI1 calls: Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1 Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1

Dear Asterisk developers, I run the same asterisk version on the home machine and on the work. On the home machine I have Slackware 10.0 (kernel 2.4.24) while on the work machine I have Mandrake 10.1 (kernel 2.6.8.1 <http://2.6.8.1>). When I run asterisk on the work machine, these warnings and error appear (there are no warnings or error at home): [ Booting......Oct 17 18:19:04

I recompiled Asterisk with the aggressive echo cancellation on. That's all I changed, honest. After recompiling, it refused to run. I tried updating the source, etc, and eventually went back to no echo cancellation. Every time, I got this error while starting Asterisk. Please help! I have no idea what went wrong. Oh, and yes, wcfxo and zaptel are loaded, I checked with lsmod. I

I am using CVS-v1-0-04/20/05-12:31:26. Windows Messenger5.1 works MOH fine. After I stop MOH on Windows Messenger, if the hungup signal could not send to *, the sip channel(e.g.,SIP/52001-08ca) for this MOH remains. Then the user trys again MOH, a new sip channel starts. And again the hugup signal can not send to *,......... When I 'stop now' from CLI> , * cleanups the remaining sip

Can anyone explain the Polycom Text Messaging features built in to the IP 500/600? Can Asterisk (or something else) talk to it? I've seen vague references to MSN Messenger, and somehow that's mentally disturbing. Chris Coulthurst <mailto:chris@shuksan.com> chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anyone else unable to get to www.voip-info.org? Site is returning 'connection refused' here. Chris Coulthurst chris@shuksan.com

Does anyone have a quick-n-dirty context to implement *66 automatic callbacks? I have a few people who like to have no call waiting on their phone (can you really blame them?) It would be nice to have something like *66, and also like 'Camp On', but instead of waiting something like 30 seconds, monitor the channel until it becomes available, then immediately ring back your phone to

Does anyone have success in upgrading Aastra/Sayson 480i CT firmware? All I get, no matter what I've tried is "Unable to upgrade firmware". tftpd is working because the dialplan freshens, and I have aastra.cfg whatevermacaddressfile.cfg and firmware.st in /tftpboot Am I missing something stupid? Is there another way to upgrade it? Chris Coulthurst chris@shuksan.com

I was copying tracks from CD into mp3 files so that I could use it in Asterisk 1.4.21.2 MOH. (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

Hello, I have Asterisk running on a RH9 box; Everything seems to be working as it should, except for Festival. Every time that Festival is called from Asterisk, Asterisk silently shuts down. Festival doesn't give any error indication and Asterisk just plain dies without a peep. Festival was installed per the Wiki, using source and patched with festival-1.4.3-diff; it works fine at the

Does someone have a link to step-by-step instructions to making the Line-In on the console sound card a MOH source? I know this has to work somehow. Chris Coulthurst <mailto:chris@shuksan.com> chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050512/4a3c3025/attachment.htm

I'm having a problem with the callerID that the polycom IP600 phones are displaying. I would like to modify the CIDName and leave CIDNumber as exactly what the phone call came in as(provided they aren't hiding callerID). Most of the calls will be going to the queue, but a few will go directly to the SIP phones. I've done a various combinations of using SetCallerID(),