similar to: The Phantom Call..

Hi, I have been trying to figure out why the S100U is not performing very well on RH9.. Here is my thinking..( may be totally wide of the mark but here goes anyway) I remember reading somwhere that the sound system used by RH has changed... Does the S100U not depend on the sound subsystem?? So what I think is that the sound subsystem in RH9 and the S100U are not happy working together.. Does

Look like SNOM have rolled back the firmware version of the 200's from 1.16w to 1.16q.. Anyone know why? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi all, Is there any alternative hardware components for multi port FXO cards, other than Single and E1 or T1 level? For example 4 or 8 port FXO card is ideal. Also the price matters. Thanks!

Hi, I have just been playing with the latest X-Lite.. It works fine with Asterisk.. As for codecs I tested G.711a/u, GSM and iLBC... iLBC is the only one that didn't work.. not sure why.. But the bigger problem is that when I call another extension that is using a Snom200 the call connects but there is no audio in either direction.. I have tried G.711a/u and GSM and while X-Lite shows that

Hi, What is the correct chan_capi dial syntax?? This is what I think it is.. exten => _90.,1,Dial(CAPI/[msn here]:${EXTEN:1}) This seems to work for local numbers.. but I have an access number for cheap long distance calls.. wich gets dialed and then the number I want to call is sent as DTMF after a few waits (w).. On my X100P I used the following.. exten => _9001.,1,Dial(Zap/1/[access

well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my

Has anyone written a script that can be used as a cron job or similar that will test if Asterisk is running and if not restart it?? I have just had an issue where asterisk crashed and someone was trying to call me.. it would be nice if it could have been automatically restarted.. I was thinking of a simple bash script something like running "ps -aux |grep asterisk" and then some kind

I am going to try and use a passive AVM fritz BRI card for my * setup.. Here is the thing.. I need to order my BRI from BT.. The service that looks to be the one to use is what they call ISDN 2e becasue this has the option to setup hunt groups across multiple ISDN2e lines so I could add another line later to get 4 channels.. According to the BT website in order to use the hunt grouping across

Hi, Does the G.729 module support adding more licences??

I know the recommendation it to not run more than 2 T/E100P's in a system but what about X100P's.. Usually there are 5 PCI slots in a system, has anyone tried 5 x X100P's in a system? Later.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi, Is Asterisk (or can it be set up as) a SIP proxy? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

I've had the same thing happen, only on the single port T1 card and a channel bank, and one of the FXO channels also having a phone attached elsewhere... I just wound up putting that channel in a different context and running Exten => s,1,Hangup (I'm just using the line for outbound dialing) -----Original Message----- From: asterisk-users-admin@lists.digium.com

Hi, I installed s fresh system yesterday and it seems that zttool did not install!! ztcfg is there.. Anyone else had this problem or is it just me? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi, Has anyone worked out a way to use Asterisk in a Hot Desk environment?? I have not been able to think of a way for the user to have control over which IP phone will ring when that users extension is dialed without the user needing to reconfigure the phone.. Something like this would be cool.. User dials *8555 (or similar) and is prompted to enter their extension and then password, after

Hi.. How do you accept user input while waiting or playing moh? My Dialplan is as follows.. ring,ring,.. Hello thanks for calling blah blah... Please enter the extention number blah blah... WaitMusicOnHold(10) If no input pass call to operator.. The problem is that the user has to input the extension while they are being told what to do.. any input during Wait or WaitMusicOnHold is ignored...

Hi, Not sure if anyone will be intersted but I have put together an install guide for Asterisk on RedHat 9.. Its nothing special but it may be of use to the newbies.. Like me.. If you would like a copy let me know and I will send it to you.. later.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi,. How can I make that Voicemail app to play only my own recorded message without the default one? Thanks, Dan

I see in the changelog that ENUM support has been added.. anyone know what this is? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi, Quick question to all the electronics gurus out there.. I unpacked my second GS phone yesterday (had it for about a month!) and set it up.. This morning the power supply is dead.. I have looked for a new one online (In the UK using Maplin let me know if you know a better place.) becasue it would probbaly take too long to get one sent from China or the US and I need to get that station

incominglimit is already implemented for SIP. Just specify under the endpoint how many incoming connections are allowed. For example, [cisco] type=friend username=cisco secret=blah nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away