similar to: Re: Voicemail problem, not playing back

Following up with some more tests... I just left a message in a voicemail box, and got the same behaviour (i.e. it says its starting to play ('First Message') then it goes to the VM ending menu (i.e. press 7 to delete) without playing the message. On this particular test.. I went ahead & left a 2nd VM in the inbox, then when I went to check them, they would both play properly... The

Hey all, am running into a problem with * 1.2.1 recently. When we leave a voicemail for someone, occasionally when they check the vm, * doesn't play back the message that was recorded. I can see the vm audio file on the server, but when it's checked from a phone, it just skips to the end of the session asking if you'd like to delete the message, even though it never played. Any ideas

Hi all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd

Hi again, I'm trying to setup our non pri T1 (they call it a Long Distance T1), our current pbx has the signaling set to E&M, I can set em in zapata.conf, but I'm trying to track down the proper entries for the zaptel.conf file. The digium docs only show a PRI example. Our current system has these settings: Signalling: E&M Framing mode: ESF Line Coding: B8SZ here's my

Anyone gotten two of the 2572 echo canceller cards to work in a 253c mounting assembly? I can get one to work, but when I install two, one always fails. I've tried all my cards solo in the enclosure, on each side, and they all work properly when only 1 is installed, however, when I install two, one of them will come up, but the other always fails. Anyone know what might be causing this?

30 says it's view only in the docs & I can't seem to change it, any other options? > Option 30 allows to set Module Shelf Address/ID.

Hey all, been noticing some oddness on a new AAH install... occasionally an incoming zap line with automatically connect with an outgoing extension, even though the incoming line hasn't specified what extension it's aiming for (i.e. haven't tapped in the ext # yet)... so someone's trying to call out from inside the office & are automatically connected with an incoming line.

Hey all again, I'm wrestling with echo problems on our sip extensions. I've set these items in zapata.conf but tweaking these values doesn't seem to make much difference echocancel=yes echocancelwhenbridged=yes echotraining=2500 rxgain=8.0 txgain=1.0 are there other settings that can help me tame this beast? Been searching but not turning up anything that'll work here. Thanks

Hey folks! For those who are interested in James Elder's publications: The links on his page work perfectly now. http://elderlab.yorku.ca/~elder/publications.html Thanks a lot, James! Everything seems to be alright now (although a little sluggish :-) Philipp -- James Elder wrote on 16-1-2001: > Hi Philipp, > > I have fixed up the links - please give it a try and let me know if

Hi all, been searching & not finding an answer to this, although I'm guessing it's absurdly simple... I just hooked up a T1 to our * box (1.2.0), which had been using POTS lines via a channel bank.. Now when I call the new T1 circuit, there are no rings, the Autoattendant just picks up right away.. Any clue on how to make it ring twice before getting picked up? I tried immedate=no and

hey all, trying to get a zap extension to work & I can dial out normally with it, but if I try to access any of the features (i.e. *97 for voicemail) the zap channel doesn't hear it, and all i get is dialtone. Is there a dialplan setting or something to make the zap channels recognize keys like * or # ? Thanks in advance

Is there an easy way to disable flash transfers? I'd prefer the users hit # to transfer, since some users are hanging up a call, then dialing another one without giving the handset enough time to actually hangup the call, so it appears that they are transfering the 'ended' call to the new number that they are calling.. I'd like to keep flash functionality for call waiting, but

Hey all, is it safe to run the asterisk-update.sh script that comes with AAH to upgrade only the asterisk binaries? Doug has chimed in a few times saying 'upgrade' when I post problems, but Aah makes this really painful. I'm using AAH 2.0 & am fighting a number of 'bugs' that only seem to be manifesting in my installation. Can I safely upgrade just asterisk and not any of

Hi all, using * 1.2.9.1 and this week all of the sudden calls keep ringing even after they've been picked up... Here's one users summary: When I pick up the phone, I hear a dial tone and I am able to dial out. But for some odd reason, the receiving line picks up while the outgoing line is still ringing. And the receiving line can hear everything while the phone is still ringing. I tested

Hello, No public anwer came to my previous post, and my problem is urgent, so I'm trying again. I'm using Ogg Vorbis since 2000 and some of my files appear to be damaged - at least some players reject some of them (WinAmp, Quintessential Media Player beta 116). Is there a repair tool for Windows which could rebuild vorbis files so that they all could be played in WinAmp, which is very

> I'm not sure exactly what it is you want to do. You say you want several > genre-based mountpoints, but you then want to randomly switch between them. > Wouldn't it be easier to just have one stream of files randomly selected > from the various genres? Or maybe I'm missing something? 1. I tried the ezstream with a m3u file and works fine. Worked after 5 seconds, but

The only thing that has changed is the winamp version number. My radio streams are unaltered (to the point of the same process connecting to two versions of Winamp). What happens is that I begin to hear skips in the stream. Soon the stream is full on stuttering, as if a square wave was modulating the volume control. Anyone else experience this? ------ Dave Hayes - Consultant - Altadena CA, USA

hello..... i am running a samba server (samba 2.2.3a on suse linux 8.0) , two win 98 client and one win xp client. the access to the samba-shares and the filetransfer works very fine with all clients. but if i want to play mp3-files that are stored on the server from the win 98 clients nothing happens...winamp just 'hangs' trying it....there is no problem with the xp client any

>>I realize that I may have unintentionally been misleading you by leaving some information out. That error report from ogginfo keeps going, repeating the same 3 errors (as above) over and over again. It ends after a stream_truncated=true, not giving a final header_integrity=fail. (Perhaps because there's not another packet?) Then it reports a total playtime of 0. > >Yes,

I have just finished setting up ices2 => icecast 2 on Suse9. Have 5 or so .ogg tracks in a playlist. Also downloaded a full copy of winamp 5 for win. If I point my browser at 10.1.1.1:8000/playlist.ogg then I get the screen to download file I dont know what is but winamp doesnot up. If I use the open URL feature in winamp and give it 10.1.1.101:8000/playlist.ogg Then winamp try to connects