similar to: Asterisk-gui

Hi list, Anyone knows about the date of the official (stable) release (v1.0) of AsteriskNOW??? It's supposed to be at the end of this year, which is very close now with no signs of it. Thanks... Raul -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071218/c543b407/attachment.htm

I'm impressed with the call log reporting that Switchvox offers. There's a screenshot here: http://www.switchvox.com/sv?cmd=screenshots&pic=13 Can normal Asterisk do this? I installed AsteriskNow on an old computer, but I couldn't find anything like that in the GUI. I found the CSV file in /var/log/asterisk/cdr-csv, but that's not much help unless there's some other

Hi folks, we use meetme application with pin so when a customer joins he's prompted for his name. Then the voice say:"press one to accept the recording..." My question is, is it possible to cut off that request to"press one"? Thanks to all -- .:FaberK:.

Hi list, I'm trying, as usual, to set up voicemail. It works, but signaling to phones, doesn't. Into XLite logs, I have: -- Messages-Waiting: yes Message-Account: sip:voicemail@mydomain.com Voice-Message: 1/0 (0/0) -- but nothing appear on the XLite screen. So, I understand that I'm able to send the right signal, but something is still wrong. Ideas? Thanks in advance -- .:FaberK:.

Hi folks, I've just updated my * and I noticed that from the update the uniqueid field into mysql, is not written and ASTPP do not charge the calls. I got an eye to cdr_mysql.c and I found that at line 212, into one insert query, uniqueid is missing. But I can be wrong. In any case, somebody got same problem? Any suggestions? Thanks to all. -- .:FaberK:. -------------- next part

Hi to all, I got a Cisco 2651XM wired to an E1 PRI. What I want to do is to pass all incoming calls to my asterisk. This is my actual conf: http://pastebin.ca/270677 with this I'm able to call my number from outside, but the call stop on the 2600, infact I can hear the tone, but I'm not able to forward calls to my asterisk. Anyone got an idea of my errors? Thanks to all. -- .:FaberK:.

Hi all! I've got a problem with thia PA168S/AT320P telephone. I got 2 servers: one with SER and the other with Asterisk. All users are on SER and Asterisk is the gateway/voicemail. In these days I'm starting some tests using Asterisk accounts users. With this PA168S/AT320P, if I use it with a user from SER, it's ok but I can forget to use it with Asterisk users!!! I've also updated

Hello Guys, first of all, I'm very new with asterisk. I'm trying to set it up. I've already compiled and installed Asterisk-1.0.7 Now I'm trying with asterisk-oh323 I've already installed pwlib, oh323 and I've already set the variables. Now, when I try to "make" asterisk-oh323 I receive this error messagge: for x in wrapper asterisk-driver; do make -C $x all ||

Greetings, Last week I began researching Asterisk for the first time. I did what most noobs would do; downloaded an image that seemed simple and straightforward and had some credibility (*now). I also downloaded the TFOT version 1 as a guide. As questions arose, I tossed a few out in #asterisckNOW channel..and found it to be a ghost town. Only later did i start to ask a few

We are in the process of preparing to move our Asterisk server to a Digital T1 interface card instead of a analog card (via an Adtran which is now connected to the T1). I did a preliminary test the other day and hooked the T1 line up to the T1 card, bypassing the Adtran. This worked rather well I must say. The two issues I ran into are: 1) Caller ID is not working even though I enabled

Hi to All, I've an Asterisk CVS Head working with Mysql. My problem is that instead of ANSWERED or something like, into the CDR database records, I find only numbers. This is also a problem to let ASTPP works, infact I receive an error: ERROR - ERROR - ERROR - ERROR - ERROR DISPOSITION NOT MATCHED and the call has no cost. Any suggestions? Thanks -- .:FaberK:.

Hi, I'm using voipbuster at work, and I've got 2 questions: 1) Is it possible to send faxes using voipbuster connex? 2) Is it possible to cut off or cover the voice that say the charge per minute?(I've payed the '5' euro, and from that moment I've got it!). Of course I understand that is to let me know how much I'm going to spend, but I do not like it, expecially when

Hi all, I've got an Asterisk box with 1 Sangoma A102 and 1 Ericsson PBX. I need to use Asterisk as E1 line for the Ericsson PBX. How do I have to connect them? I'm trying to connect the Sangoma to the Ericsson, but RED alarms remain. Any suggestions? Thanks -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guys, I'm using Zyxel Prestige 2602R, as router/SIP-ua with my architecture SER+Asterisk. Normally, everything is fine. In these days I'm experiencing some problems: some guests said me that, if he call everything is right, but if is called, he cannot hear the caller. Immediately, I though into an RTP-Proxy problem, but is not. Then I saw that message appear on the Asterisk CLI, during

Panasonic PBX KX-TDA600 it doesn't support SIP protocol for VoIP technology it only support H323 Trunk through 4 or 16 channels gateway card and TDM technology with ISDN BRI and PRI card. Mc GRATH Ricardo

Hi All; Is there an GUI for Asterisk that can help in showing the call flow (who is in progress, who is connected, called number, ...)? I was think in AsteriskNow does this? Any advise? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

Hi to all, I got an Asterisk with 2 BRI(7 pstn numbers and 4 concurrent calls) and 15 SIP extensions. The receptionist has a SNOM-360. How many SIP accounts would you configure on that phone? Only one would be enough? One SIP account, has a limit on concurrent calls? I saw that the SNOM-360 can handle up to eleven SIP accounts. Thanks -- .:FaberK:.

Hi guys, maybe you?ve got the answer...! When a caller(not internal, but from PSTN) call *, I need to let him hear a message, before * answer and the bill start running. If is not clear, just let me know. caller->telco(telco bill to the caller as soon as * answer)->asterisk Thanks in advance. -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi folks, I'm trying to install asterisk with radius cdr support. I got freeradius up and running, so following radius instructions inside asterisk source package, I've installed radiusclient-ng and relative headers. But when I start configure(asterisk 1.4.18.1) I got: checking for rc_read_config in -lradiusclient-ng... no If I type: ./configure --with-radius=/usr/share/radiusclient-ng the