Re: [Asterisk-Users] voicemail-hangup issue

Yes!, The latest CVS has fixed this problem. Thanks for the help. Sean - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, April 06, 2004 6:53 PM Subject: Re: [Asterisk-Users] voicemail-hangup issue I have a small * installation with 2

Re: [Asterisk-Users] res_motv: Request for Comment

I wouldn't want a call home feature that is enabled by default. I think it would be great though if * had some ability to update itself. Perhaps via a CLI command, as others have suggested. Something similar to RH's up2date would be great in my opinion. Anyway, thats my 2 cents. Sean Rodger

Re: [Asterisk-Users] errror compiling asterisk from cvs

I am getting this too under RH9. Sean - Original Message - From: Alessio Focardi [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, April 07, 2004 9:04 AM Subject: [Asterisk-Users] errror compiling asterisk from cvs I got this compiling the new cvs code ... any idea ? Tnx

[Asterisk-Users] voicemail-hangup issue

I have a small * installation with 2 incomming analog lines connected to 2 X100P's, and several SIP phones. With the most recent update (cvs-04/01/04), I have started to see a problem. If someone is connected to voicemail, and hangs up without leaving a message (the problem does not occur if a

[Asterisk-Users] calls dropped with grandstream

. Does anyone know if there is a way to fix this? Eventually I will replace the Grandstream, but for now I would just like to fix this problem. Sean Rodger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo

[Asterisk-Users] RE: Re: calls dropped with grandstream

I have firmware 1.0.3.81 I have a Cisco ata186, and that seems to work fine. Infact, I've had that up for about 3 months without rebooting, and it is still working great. The dropped call problem only happens on the grandstreams. Sean Rodger

[Asterisk-Users] ISDN newbie

Mine is a small company with 2 incoming analog lines that we use for voice. One line rolls over to the other if the first is busy. I started using an */grandstream combo a while ago, and besides a couple of bugs that I have yet to work out (echo, ringing in the earpiece) its has been good for the

[Asterisk-Users] grandstream ntp

I am running ntpd on the same machine as asterisk in order for the grandstream phones to display the time. After a while the time display fails until the phone is re-booted. Has anyone run into this problem before? Is it simply a bug in the GS firmware? Sean

[Asterisk-Users] Re: grandstream ntp

minutes, hours, days... ? It happens after about 6-7 hours. The problem is very consistent. I am also running 1.0.3.81 firmware on the phones. Perhaps this is a config problem with ntpd? here is my ntp.conf: server clock.isc.org server time.nist.gov restrict clock.isc.org mask

[Asterisk-Users] critical problem

About every 10th call coming into my x1000p is not getting the audio it should. You can see the messages scrolling on the console as they usually would, playing the thankyou, then and menu messages. internal phones ring, but when answered there is no audio. The caller gets a full volume echo

[Asterisk-Users] Re: call waiting beep

I am thinking of coding a solution using variables, Cut, and ChanIsAvail. here is what i'm thinking of doing Create a variable that contains the string SIP/gs1SIP/gs2SIP/gs3 ... etc check each phone with ChanIsAvail, and use Cut to remove its representation in the string (if its not avail) then

[Asterisk-Users] call waiting beep

,25,t) If one of those lines is being used, then the user gets a really loud call waiting beep, and on the ata186, also an inband callerid noise (perhaps changeable on the ata186). thanks for any info. Sean Rodger ___ Asterisk-Users mailing list [EMAIL

[Asterisk-Users] my asterisk experience (long)

I thought I'd post my experiences for the benefit of anyone else who may be at the point I was when I first started with asterisk. I have 2 incoming analog lines (north eastern U.S., Verizon) where one is set to ring if the first is busy. I bought a bare-bones system from abs-pc with the

[Asterisk-Users] Re: TDM400P??

Here is some more information about my problem: With 2 phones plugged into the 4 port FXS card, here is a situation I have witnessed: I have a clean dialtone one phone. The instant the other phone goes from on-hook to off-hook, the clean dialtone on the first line turns into a loud crackling

[Asterisk-Users] Re: TDM400P??

So, do you have the P/S 4-way connector plugged into the TDM400P ? Regards...Martin Yes, and I've tested the voltage to the card. Both the 12V and 5V supplies are OK to the card. -Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] TDM400P??

Here is my system: POWER SUPPLY 450W ALLIED ATX450P4 R(41) MB NFORCE2 A7N8X DELUXE ASUS RTL(Standard) CPU AMD|2500/333 ATHLON XP BARTON R(Standard) DDRAM 256M|DDR333 PC-2700 -K %(Standard) HD 40GB|WD 7200RPM 8MB WD400JB%(70) VGA ASUS|V8170MAGICII/T 64M MX440SE(58) CD ROM 56X|AOPEN CD-956

[Asterisk-Users] tdm40b

I have 2 xp100's and one TDM400P. I've plugged a phone into the tdm40b, and when i take it off hook sometimes i get a dialtone, other times i get the message Power alarm of module2, resetting spit out to the console from the wcfxs driver does anyone know what this could be? I've tried two

[Asterisk-Users] Linux flavor?

What Linux distribution is best for use with Asterisk? (easiest compile, least problems, etc) Thanks, Sean Rodger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users