similar to: MOH failures (bad quality with interruptions)

Heya list, I need some advice/experience. Some of our customers are asking us about DECT solutions for their asterisk install. Some others will not go to asterisk if there won't be a DECT solution. They now have a Siemens or a Samsung PBX. Those PBX-es come with a DECT basestation and optionally repeaters etc. All those basestations speak some own protocol to the PBX, so we cannot use them

Hello, I am a new R user having transitioned over from S-plus recently. I have a question that is probably very trivial but I am having trouble finding a solution. In S-plus, graphic pages are created as tabs when multi-page figures are created. I have shown the R code for xpose.VPC (a function within library xpose4 for R) where I want the figure from each Strata (STRT) to displayed on a

Well, I wake up this morning, and aussievoip isn't up. I ring godaddy, who _were_ hosting it, and they say that the machine's been compromised, and you can't have your data. Nyah Nyah. I spent 1 hour and 38 minutes on the phone to them, trying to convince them to let me somehow get access to it, but to no avail. I've reported it to the Australian Federal Police High-Tech Crime

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).

I've finally got a chance to play with 1.0.1.9, and the wiki has been updated. At the moment, I don't know of _any_ bugs with it. I'm yet to play with complex things like early dial, and will update the wiki as I find information. http://aussievoip.com.au/wiki-GXP-2000 I've just been given official permission to offer the firmware as a download, so if you visit

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,all I am newer to Asterisk.My Asterisk version is the newest CVS-HEAD.now something appears in the console CLI like below these,I don't know what's happen to my Asterisk Server.Could anybody help me? Thanks Junk at the beginning Warning, flexibel rate not heavily tested! Junk at the beginning Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested!

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I just noticed that the Skype API for linux seems to be available. I've read before a number of posts where people were talking about implementing a chan_skype with the skype API. I wonder if there is any progress in that direction, and if anyone is working on it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to

Who mentioned fax? :-) Native CAPI isn't an option for us (unfortunately), as our service is a point-to-point service, which the AVM CAPI drivers don't support. We've now got another problem -- we're now able to make calls -- once. The kernel panics as soon as someone terminates a call (and, so it would seem, at various other times too). This has only been occuring since mISDN

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 just can't seem to get the MusicOnHold function to work out ok. I' managed to get the MP3Player app to work out fine, but when I run the MusicOnHold all i get is siliece. I can see that Asterisk executes mpg123 properly (I think) #ps axuww|grep mp gk 4383 0.0 0.4 3736 552 pts/4 S 15:06 0:00 /usr/bin/mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 sample-hold.mp3

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

Hi ! :) I'm having trouble with my MusicOnHold... I'm using standard files (fpm-*.mp3), which are provided in * package (so I assume they're not variable bitrate encoded... anyway, I tried re-encoding them and no changes). The music is played, but with micro-cuts every 2/3 seconds... it's happening on every channels (sip, capi). So, if someone has an idea... :-) (if you want to

Hi, My * server is running FreeBSD. The wonderful isdn4bsd with chan_capi port by hselaski works great, everything is perfect in a perfect world... but ! These [@$!#] MSNs don't work at all ! My extensions.conf is very simple for my 5 MSNs (0353, 0416, 0618, 0619 and 0620) : [rnis-in] exten => 0353,1,Answer() exten => 0353,2,Dial(SIP/600,,m) exten => 0353,3,Hangup exten =>

Type 'mpg123' at the Linux CL. (no quotes) If the version is anything other than 59r, you problem is solved. Go to the Wiki and search for Music On Hold. Do the install of version 59r ONLY as described in the docs. Cheers, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Neil A. Hillard

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