similar to: WebVMail Woirks but No Audio

I am having trouble locating webvmail on my * server. Is this a seprate porgram or does it come with *. I am running version asterick*CLI> show version Asterisk CVS-03/26/04-17:08:20 built by root@localhost.localdomain on a i686 running Linux asterick*CLI> Thanks Kurt __________________________________ Do you Yahoo!? Yahoo! Photos: High-quality 4x6 digital prints for 25¢

I did a make webvmail and I get the following error on redhat 9.0 No HTTP directory make: *** [webvmail] Error 1 I have the perl-suidperl rpm installed and apache installed Thanx. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050501/275c1e94/attachment.htm

Hi, I wouldlike use webvmail on my asterisk, I use debain Sarge with asterisk 1.0.7 package. I have installed package "asterisk-web-vmail" but when i go to http://MyAsteriskBOx, i have a page of presentation of Apache. Could you help me please? Thanks

Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk

Hi Everyone, I read through the list on the issues with the ztdummy driver which I need for MeetMe, but I seem to have come across a problem that I cannot seem to find an answer for. I am running Gentoo 2.6.10 on an Intel box. I have read the the wiki entries on the ztdummy and followed the instructions as they relate to the 2.6 kernel. Everything compiled great, but a modprobe ztdummy

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040112/a6f17a16/attachment.htm -------------- next part -------------- Hello I found related question on vmail.cgi in the mailing list but that didn't answer my question. I did copy the vmail.cgi to /var/www/cgi-bin/ but still gets the following error message when I access

I have a [start] context that all my inbound and '0' calls are routed into. Because of the way I want to set my system up, I want to prompt the user to enter a 1 if they know the extension, or a 2 for a directory and nothing else. It works, however there is a 5 to 10 second delay after enter the 1 or 2 before the system responds. I have read over the wiki on how asterisk handles digit

Hi Everyone - I just got my hands on a Cisco 7970 and was told that I should do a factory reset before trying to configure it to work with Asterisk. After the factory reset, it will not boot at all, instead sits with the line button lights flashing one at a time in sequence. I have had no luck trying to figure it out - anyone run into the same problem that can lend a hand..? Thanks

>how does the phone know where to find the TFTP server..? Dude, option 150 in your DHCP server: http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186 a00800942f4.shtml We use the same option for our Mitel phones. HTH.

Hey Everyone - I was going to create a visio diagram outlining how my extensions will flow out. I was just wondering if anyone on the list may have an example they have already done up so I can get some ideas. Thanks ****************************************** Richard J. Sears Vice President American Internet Services

Guys. Anybody has developed and asterisk brochure for commercial purposes (consultant, etc) that I might be able to take a look at?

Hey Everyone, This is not an operational issue, and to my knowledge only effects the look of the command, but when I issue a "sip reload" then a "sip show peers" I see all of the actual usernames I have assigned in my sip.conf. However, five minutes later I reissue the sip show peers and all of the usernames have disappeared and are replaced by the sip ID. The only way to get

Have Asterisk us at running fine, but have run into a small snag. It's not emailing the voicemails to the users. I have attach=yes set, sendmail is configured and works from from the commandline (sent an email to myself). Unless I'm wrong, or missing something, asterisk is configured by default to send an email when a users receives a voicemail, correct? Thanx A

Our main asterisk box peers with that of a customer. We are trying to assign DID's to their extensions but get this error. What are we doing wrong? Client side Mar 25 18:49:47 NOTICE[1369]: chan_iax2.c:6545 socket_read: Rejected connect attempt from 203.xxx.xxx.16, who was trying to reach 's@' Our side Mar 25 18:56:15 WARNING[705]: chan_iax2.c:5546 socket_read: Call rejected by

Hey Everyone, I am using NuFone for 866 inbound service and I am trying to figure out the callerid part of it. Any call into my * system just shows "Toll Free Call" and will not give me the calling party's caller ID info. Is this just something I have to live with using NuFOne, or did I miss some type of config in * that will grab the callerID other than the inbound 866 number...?

How do I route all the outgoing calls through a SIP gateway, this should send more than one outgoing call to the sip gateway at once. please show me the sample configurations on how to do this. my SIP gatway can accecpt direct IP traffic or SIP proxy traffc. Thank You Kanishka -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have let's say a reception that is comprised of 2 zap extensions and a mobile phone to dial using ISDN through Capi. I want to have a delay before starting dialing the mobile phone so that it rings only when the call has been unanswered for say 25 seconds. I tried to use Capi/2106994444:ww6935555555 but without any success. There is any way to do it or the code has to be modified ? Thanks

Hey Everyone, I am playing around with my * box, and I have a few different phones hanging off it it right now. I have a Cisco 7960 capable of g729, ulaw and alaw, I have a Cisco ATA186 with a Panasonic cordless phone attached to it, I have a Digum IAXy with a dumb analog phone attached to it, and I have a Linksys PAP2-NA with an AT&T 959 analog phone attached to it. I also have several

For anyone playing around with IAXy(S100i) devices, I am making the following available: Windows IAXy Provision v1.00 This is a from-the-ground-up development of a means of provisioning IAXy devices using a Windows environment. For some users, being bound to Linux for IAXy provisioning is not viable or convenient in some cases. This application provides a GUI data entry for the various IAXy

Hi, I am implementing T.38, and finding a problem getting boxes that work with T.38 for testing. A lot (maybe most) ATAs now claim to support T.38, but I'm finding a lot of these lie. I have one box here that just crashes when it hears a fax tone. :-) I'm looking for boxes known to implement T.38 properly, and which really work in the real world. Regards, Steve