similar to: enter menu

Dear all I have now install TE210P 2 port E1 card on asterisk 1.4.10 on centOS 5 but thing is that i have connect 1 E1 port with avaya E1 back 2 back and second E1 card on Direct Telcom for outgoing for outside now i got this error when i call on avaya PRI asterisk think PRI_CPE and remote end also CPE i have configure /etc/zaptel.conf span=1,1,0,ccs,hdb3

Before an IP Phones can be registered to an Asterisk server, the extension for it must be configured in Asterisk. Usually, Asterisk adminintor must add the extension by hand. Is there any library, API to do this by software??? For example, i want to develope a software that add new extensions to Asterisk system, sothat, any IP Phones can use that extensions to establish a call. I'm digging

Hello, I am a small business owner in need for a solution that automatically answers an incoming call, prompts the caller via touch-tone menu ("press 1 to leave a message, press 0 to speak to a representative") and will ring my (real) phone ONLY if requested by caller. I know that Asterisk is capable of all the logic behind what I described above. However, I couldn't find a

What does the following EDAC problem means? The machine is a AMD 64bit box running Centos 5. It looks like some problems aroung AMD DRAM Memory controller. But what does it really mean b/c most of my AMD boxes has these messages in /var/log/messages. Please help. ... Aug 1 23:29:40 ccn128 kernel: EDAC MC: Ver: 2.0.1 Jun 12 2007 Aug 1 23:29:40 ccn128 kernel: EDAC MC0: Giving out device to

How are the screenshots at syslinux's homepage taken? I would like to take some of my extlinux setup for my usb to archive as well as show to my friends. Also for the pictures under syslinux-3.51/sample/ "syslinux_splash.jpg" and "m16-640x640-syslinux.jpg" could someone edit the text for it to say extlinux instead(or the original pics)? also any other places to find cool

Hi all, Iam using asterik 1.4.8 and connected to google talk. When iam calling from my google talk account to sip phone i can hear the voice (2 way). (this happens only within the LAN). when my friend tries to call my asterisk server (connects to the public ip) using his googletalk client it comes to my sip phone but either party cant hear a voice. I have fully allowd both tcp,udp on my

Dear All I am new in this list right now i am working on asterisk server and deploying asterisk PBX in my organization now i have alread setup Avaya PBX and i want to intergrate my asterisk through mediant 2000 [asterisk]-----[mediant 2k]--------E1-trunk------[Avaya] this is my setup now i want to create dialpan so how to forward call in to existing avaya setup means i have not

retreive the variable and did not write it directly ${CONSOLE} as already CONSOLE is configured in the [global] or what is the storey :) - ? Regards, ---------- Bilal Ghayad ____________________________________________________________________________________ Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. http://tv.yahoo.com/collections/222

dear all is there any GUI application with support asterisk 1.2 version i am useing 1.2 and i have fine more about GUI base configuration but i didnt got any GUI package for asterisk 1.2 --------------------------------- Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. -------------- next part -------------- An HTML attachment

Hi: Which was released for free download under a Creative Commons license for "The Future of Telephony, by Leif Madsen, Jared Smith, and Jim Van Meggelen". Regards. --------------------------------- Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear List; To have better security, how can I put a password on the international calls (if the user dialed the international call, then it will be asked for password to send the call outside)? Can this password read from the CDR file to know whom did these international calls (using which password? As I might have mutliple passwords). Regards Bilal

$ cd /tmp $ echo "123" > aaa $ echo "123" > bbb $ ln -s bbb ccc $ ls -la aaa bbb ccc -> bbb $ rsync aaa ccc $ ls -la aaa bbb ccc This example shows that symlink ccc was replaced with a regular file, which implies that during a remote sync the content of the aaa will be downloaded. However, files aaa and bbb have the same content, so I need rsync to skip updating the

Dear Philipp; Kindly find the part of the configuration as below: [general] allow=all disallow is comment by ( ; ). [bilal_sip] type=friend context=internal host=dynamic canreinvite=no dtmfmode=rfc2833 So where is the problem? The endpoint does not register and nothing appear on trace level 3. And the amazing thing that if the endpoint send wrong username (for example: bilal_sip100) then it

Most of my website is static HTML, but I do have a couple PHP pages. I''d like the default template to be applied to it (i.e. I don''t want it to just be copied). How can I get webgen to create the PHP file with the right extension in the output directory? A way to specify the following mapping would be nice: *.php.page -> *.php *.page -> *.html Thanks. -

Hello list, i need help. My problem is that when I want to call (using ISDN phone or internal SIP client) via the Sip provider a real phone number (ISDN phone or internal SIP Asterisk >> SIP ), I get a ring tone. When I now decide to hang up (e.g. if nobody answers), the called telephone continues to ring almost forever. the sip.conf: [2563105] accountcode = 2563105 username =

Hi: I've configured an asterisk server with A102d sangoma's card and the E1 link.I want to dial outside number using the E1 Link.How can I dial a phone number?Is this true? exten => 123,1,Dial(ZAP/1/phone number) I'd appreciate any help. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi: I configured an asterisk server with 2.4G cpu and 1G ram for conference call service but when 5 peoples about 20 minutes were talking together suddenly asterisk was disconnected.May it has happened because low cpu or ram?I saw var/log/asterisk/messages file but everything was going well apparently,asterisk was disconnected suddenly.What 's  your idea?Please  guide me.

Hi: I want to have conference call with asterisknow and need 2 ports E1.Which Digium card is better?TE212 or TE220.I haven't problem with motherboard. Regards. --------------------------------- Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi: Please every that work with A102d say how about is it?Is it really difficult to install card for me new in asterisk? Best regards. --------------------------------- Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, quick question from a social scientist new to R: how do I add labels/row names to the nodes in coordProj / plot.Mclust plots? Apologies if this is a silly question... Thanks. -- Bobby ____________________________________________________________________________________Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when.