similar to: Voicemail Question

After rebooting my asteriks server, e-mail notifications are no longer being sent after a voice-mail is left. I can see the messages in /var/spool/asterisk/vm. has anybody had the same experience? how was it resolved? Uri -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040410/fc494bb4/attachment.htm

Hi, I run a local exchange server and would like asterisk to send voicemail notification messages via exchange. I have had a look at the voicemail.conf file, but I can't see how I would go about configuring it to use an account set up on exchange ? The exchange account would have both POP3 and IMAP access, so how can I tell Asterisk to use the exchange account rather then sendmail ?

What is the proper method to install/configure an X100P FXO card? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031009/ce8a7803/attachment.htm

Stutter tone when voicemail in boxHi, I'm having a problem getting Asterisk to make a stutter tone when a voicemail is received. It used to work, but since I rebuilt my server I can't get it going. I've set put a line in the sip.conf file: mailbox=100 to check it, but it's not doing the stutter tone ! I am using the stable version of asterisk from the CVS tree. Any help /

Hi guys, I know this isn't strictly about Asterisk, but it is related... I am looking to buy a few IP phones, but I don't have a huge budged (hence why I love Asterisk, its amazing and free !), so I was wondering if anyone knew where I could get some cheap IP Phones ? Ideally they should be no more then about ?50 ($90). Thanks, Paul.

Hi, Does anyone know where I can get a telephone card that will fit into the PCI slot on my PC and work with the UK telephone system (BT) ? I would really like the retailer to be based in the UK if at all possible ? Also, is there any way to set up asterisk so that only certain phones can make external calls, but all phones can receive incoming calls if they are routed to that phones by some

Hi, I have got one X100P telephone card in my Asterisk server and it's working well. I have two phones lines and would like to install a second card so I can use both lines. I have installed the card and tried to set it up, but all to no avail ! Could someone outline the changes that I need to make (and in which .conf files) in order to get the second card going ? Thanks in advance,

I tried it w/ mine as well and it hung up on me because I just have Voicemail running not Voicemail2. It seems as though you have Voicemail2 because it's trying to play the Unavialable message. Just a thought though. Does it do the samething w/ [qout-phillyq] exten => 0,1,Voicemail(u1) exten => 0,2,Goto(default,s,1) Tim Thompson http://www.amatechtel.com (806) 722-2227

Hi, When I installed my X100P card there was a two ring delay before the phone started to ring. I managed to solve that by stopping the Caller Line ID being passed to the phone. I'm in the UK so that doesn't matter anyway. The problem that I have now is that when a call comes in on the external phone line and it is answered by a standard phone (not connected to Asterisk), the IP

Hi, I have just upgraded my Cisco 7960 phone to SIP firmware today and I have to say it's working great with Asterisk. At work (which uses Cisco Call Manager), when a voicemail is recieved the read light remains lit until the voicemail is retrieved. Is there any way to achieve the same effect with Asterisk ? Thanks, Paul. -------------- next part -------------- An HTML attachment was

Hello all, I haven't used cisco phones as yet - do they work with asterisk, are they good which models are the best? I am after a starting point! Thanks Nick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040422/f5246238/attachment.htm

Hi, I have a problem with my Grandstream phone. I have set it up to use DTMFMODE=info and I am able to transfer calls that have been made from that phone, but I am unable to transfer calls made TO that phone ?? I have tried every conbination of T and t in the extensions.conf file, but all to no availe ! Can anyone help ? Thanks, Paul.

Full_Name: John C. Deva Version: 2.8.1 OS: Fedora Linux 8, 64 bit Submission from: (NULL) (193.200.150.189) I notice that it is possible to redefine 'if' as a function of an arbitrary number of arguments. Such redefined 'if' can then be used as any other user function, except for that the parser still demands exactly three arguments to be given to if. Furthermore, even if

Hi, I downloaded Asterisk using this command a couple of weeks ago... # cd /usr/src # export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot # cvs login # cvs checkout zaptel libpri asterisk Can someone tell me what I need to type in order to get the latest stabe version when I rebuild my server ? Thanks, Paul. -------------- next part -------------- An HTML attachment was scrubbed...

Does anyone know if you can put two of the X100P cards in to the same machine and have access to two landlines ? I just need to know if it's worth buying two or not ! Thanks, Paul. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040410/2608a3fd/attachment.htm

Hi, This might not be a very popular question, but I was just wondering if anyone have ever tried to run Asterisk on a Windows computer using Microsoft Virtual Server (http://www.microsoft.com/windowsserversystem/virtualserver/default.mspx). I am told that you can run Linux on a virtual server using this software, so in theory it should be possible. I run a windows based domain, but am also

I tried to encode two minutes from an AVI DV file 720x480: ffmpeg latest release produces a FLV 320x240 25fps 360k video 40k audio in about 45 sec. ffmpeg latest release produces an OGG 320x240 25fps 360k video 40k audio in about 1:20 sec. ffmpeg2theora the same in about 2 minutes or more. The FLV quality it's quite good, some dropped frames maybe. I was wondering to know if ffmpeg team

Hi all, After a long time having no time, I have finally done some fresh work on my software fax machine. I have replaced the original carrier tracking with something more robust. I have also added 4800, and 2400 bits per second modes, and cleaned up a few bugs in areas like superfine mode operation. I apologise for this update taking so long. At ftp://ftp.opencall.org/pub/spandsp you will

Hi, I have just enabled call parking on my Asterisk system and it's working well. However, I am running some Grandstream phones on my system and when you press # on them to transfer the call, the user on the other end of the line hears the tone produced by the phone, rather then Asterisk recognising that it's a command. Has anyone else has this problem and if so how did you correct it ?

Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com