Re: [Asterisk-Users] MOH/mpg123 broken when running asterisk as non-root?

Marconi Rivello wrote: mpg123 writes to standard output... no ports. But, the suid might actually work. chmod +s mpg123. It makes mpg123 run as the owner (if you set the owner to be root, then it will be run as root). Marconi, thanks a lot. That did the trick I just did a: chmod u+s /usr/l

Re: [Asterisk-Users] MOH/mpg123 broken when running asterisk as non-root?

Marconi Rivello wrote: Does MOH work if you run everything as root? I mean, have you actually tested? I'm asking becaus MOH didn't work in my install, and I found out that the mpg123 in my distro was, actually, mpg321. And the parameters necessary weren't available... If this is the case, get the o

Re: [Asterisk-Users] MOH/mpg123 broken when running asterisk as non-root?

doesn't mpg123 bid to some port? try to make it suid maybe... Not positive, but this sounds right. What exactly do you mean by 'make it (what?) suid' ? Some clarification would be helpful. Thanks, jl ___ Asterisk-Users mailing list [EMAIL PROTECTED] h

Re: [Asterisk-Users] MOH/mpg123 broken when running asterisk as non-root?

Steve Maroney wrote: Can non-root users play mp3's with mpeg123 ? It may be a permission problem with your audio device files. Pretty sure that's not the problem, b/c if I switch to my non-root user and fire up the mpg123 player, it goes off without a hitch, and will play an mp3 (I don't have spe

[Asterisk-Users] MOH/mpg123 broken when running asterisk as non-root?

Hi guys, For the first time, I'm attempting to run asterisk as a non-root user for all of the obvious reasons. I'm attempting this with asterisk-1.0-RC2, based on the fairly straightforward directions found here: http://voip-info.org/tiki-index.php?page=Asterisk+non-root The only proble

Re: [Asterisk-Users] Hook-flash timing

Finally, can I turn off the '#' to transfer, since we're using the hook-flash (albeit manually) instead? ISTR an option to do this but have spent the morning trying to find it again unsucessfully... I think you might want to look at the 'T' and 't' options on the Dial application, documented somew

Re: [Asterisk-Users] sidetone noticeably loud on analog handsets on T100P

I figure it probably must have to do with my Rhino Equipment FXS channel bank guys... I'm in the process of researching if there's anyway to adjust it on the equipment. I also agree with what some of you guys are reporting re. the rx/txgain not making any difference. I think that's for the s

[Asterisk-Users] sidetone noticeably loud on analog handsets on T100P

wondering if there's some way I could be adjusting the sidetone in Asterisk or should I be looking at my FXS channel bank? Thanks, John Lawler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users T

Re: [Asterisk-Users] Asterisk -- PBX Do Not Disturb

You don't have to put this in the dialplan. It's one of those low-level functions in Asterisk (possibly controlled at the driver level-- I'm not sure about that). If you have an extension defined, pick up the handset and dial '*78', you should see on the Asterisk CLI: Enabled DND on chann

[Asterisk-Users] zapbarge w/o the mute

I've got a couple of different situations where I'd like to do something like zapbarge into a specific channel but I'd like to be able to actually talk to the party or parties on the channels, not just listen like w/ zapbarge. There are two scenarios I can think of right now where it'd be very

[Asterisk-Users] * dialing before line is open?

Hi guys, I've had a sporadic problem recently with one of my users on our POTS line. About 1/3 of the time he dials a number (usually from a speeddial on his phone, I think), he'll get some phone company message (from the outside) about how the call could not be completed as dialed or somethi

[Asterisk-Users] yet another question on DID trunks

Hey Steven, Sorry to bother you yet again w/ a question on my seemingly endless quest to get DID trunks setup for a customer. If you don't know anything about this issue or would rather I looked elsewhere (including the Asterisk list, I suppose), please just let me know right off the bat. I'm

Re: [Asterisk-Users] fax detection: false positive

Thanks guys, that did the trick. Tilghman Lesher wrote: On Friday 26 December 2003 13:42, john lawler wrote: Hi guys, I just moved from Asterisk release 0.5.0 to CVS 2003-12-22, and after overcoming a few changes in my configuration, I encountered one problem that I couldn't shake tha

[Asterisk-Users] fax detection: false positive

Hi guys, I just moved from Asterisk release 0.5.0 to CVS 2003-12-22, and after overcoming a few changes in my configuration, I encountered one problem that I couldn't shake that was working fine in 0.5.0. It's the fax detection. I just have a simple extension setup like this: exten =>

[Asterisk-Users] DID trunks -- equipment requirement

o help with customer education so much. As always, I appreciate all of your expertise and patience with me and the other new guys. John Lawler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] modem data calls through FXS / FXO digium cards failing

Hi guys, I think I posted on this issue before, but didn't get a response. I've still not been able to resolve the issue. I've got a small installation of Asterisk running one 4 port FXS Digium card and 1 FXO Digium card. I'm having difficulty routing modem call through one of the extensions

[Asterisk-Users] next stable release?

Hi guys, I've been running 0.5.0, which is dated sometime in September of this year and I've noticed a couple of new features in more recent code that I'd like to use, but am hesitant to go w/ CVS code. My system is not exactly a production system, it's mostly test, but I'm still leery of the

Re: [Asterisk-Users] Caller ID from Database

Guys, You can do the same thing w/ the builtin application LookupCIDName. That's exactly what it was designed for. You just store the information in database family 'cidname' and use it the same way. Search the archive or google for examples. jl Dan wrote: Hi, - Original Message -

[Asterisk-Users] prob. w/ data (modem) calls through Asterisk

I'm having major problems routing a modem (data) call through my Asterisk box. I've got a single incoming POTS line through my X100P, and a few extensions (and modems) plugged into the ports on my TDM400P. I've been using the system for a few weeks for voice applications and everything is work

[Asterisk-Users] telco access ?s -- PRI, T1, POTS?

Hi guys, I'm new to the telco game and still pretty new to Asterisk, although I've been using it for a couple of months now and like most of what I see. At my office, we've got a small two extension setup w/ two Digium cards for a single FXO line and three FXS extensions, but I'm also working

[Asterisk-Users] dialing 8 in VM2 causes channel lockup?

Hi guys, I'm running Asterisk-0.5.0 and accidentally stumbled on this problem while in the VoicemailMain2 application: If you login to it, or even if you call it w/ 's' to skip the login and press an '8' near the beginning (and possibly at any point, I'm not sure), the channel seems to lockup,

[Asterisk-Users] new voicemail notification by calling #?

Is there a place to configure the volume of the recording (which seems normal when I listen to it directly through asterisk (on a handset)) other than fiddling w/ the gain on my FXO card? Thanks, John Lawler ___ Asterisk-Users mailing list [E

[Asterisk-Users] auto 'modprobe wct1xxp' on startup?

Hi guys, Thanks for your answers on my two questions yesterday. That's exactly what I was looking for, sorry for not noticing it myself, but I'm still getting acclimated to Asterisk and even Linux--from what I see so far, I love it. I've got another one now. Since my Asterisk install and con

[Asterisk-Users] direct-inward-dialing (DID)

tly connect the DID dialed numbers and route the others to an autoattendant for extension dialing. Thanks, John Lawler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] callerid name modification (or adding)

Is there any way to take an incoming callerid string and remove the given "name" part of it and replace it w/ something arbitrary, or add to a blank name string (possibly by looking up the number in a database)? Thanks, John Lawler __

[Asterisk-Users] problem w/ musiconhold & mpg123

sp: No such device Anyone have any ideas on this? Thanks, John Lawler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users