similar to: Voicemail issue

Hi, Is there and trick to getting a conference room up and running.. I have 'conf => 7500' in the meetme.conf file and 'exten => 7500,1,MeetMe(7500)' in the extensions.conf file (in the same context as my phone extensions).. When I dial extension 7500 I get the voice saying "That is not a valid conference number, Please try again.." <beep> <beep>

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

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, 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

My system seems to be generating a call on its own... Unfortuately I can't give much more information.. I have an X100P and an S100U.. My Modem and the X100P share a common line.. When I am on the internet (which is most of the day) * just sits there and does nothing (apart from when I am testing ideas for the dial plan), but at night when I am sleeping and the modem is not connected then

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

Hi Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten

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...

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 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, 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

I thought I would give RedHat 9 a try with Asterisk..I thought it would be a good idea to use the latest version.. Zaptel, Zapata and Libpri all appear to have compiled sucessfully.. But.. (Why is there always a but??) It seems Asterisk is having issues with 'termcap' or 'tgetent' whatever that is.. Here is the output from 'make install'.. --------Start-------- if [ -d

OK I did some researches and tests with it, and finally: I registered my messenger to the asterisk and called if from the snom. The audio from the snom to the messenger was PERFECT. By the time of the call This message was running on the asterisk console: WARNING[16400]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames My conclusion is that the snom100 utilizes MSGSM

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

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, I have just got my SNOM 200 to start doing some real testing with *.. I am trying to use the GSM codec but the quality is really bad, Is that normal? does anyone actually use GSM?? Also are there any 'gotcha's' that I need to look out for so I don't spend hours trying to get somthing working that really doesn't work anyway.. Thanks.. later.. --

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

Hi, I have more or less decided to do with the SNOM phones for the next stage of testing with Asterisk becasue they seem to be the best value for money and have support for the GSM codec and easy upgrades.. But now I have to decied wheather to get the 100 or 200 and if its the 200 then I need to have some justification for the extra cost.. Can someone who knows these 2 phones tell me what the