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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

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

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

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

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

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

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:

For user and administrator enter menu when *-key is pressed we should use 's' option or nothing(asterisk does it automatic). I'd appreciate any idea. 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:

$ 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

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. -

Hi List; [incoming] include => parkedcalls exten=103,1,Dial(SIP/Bob,,tT) exten=104,1,Dial(SIP/Charlie,,tT) When we use tT and when we use t alone or T alone, I know this for call parking, but I do not know what the tT does? Regards Bilal ____________________________________________________________________________________ Sucker-punch spam with award-winning protection. Try the free

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 List; I need to use an prepaid billing system with Asterisk, and I do not know which one is more stable and integrated with Asterisk? A2Billing or AstBill or ASTCC? Also, from where I can download it and ready about its configuration? Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460

Hi List; I have this example for Macro and I am not able to understand some line, if any one can help me plz :)- [macro-voicemail] exten => s,1,Dial(${ARG1},20) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) exten => s-NOANSWER,2,Goto(incoming,s,1) exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}) exten => s-BUSY,2,Goto(incoming,s,1) exten

Hi List; I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? Regards Bilal ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.

Hi List; I read the following sentence: "The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable" In the following link: http://www.voip-info.org/wiki/index.php page=Asterisk+CLI+prompt The question is: what is the ASTERISK_PROMPT UNIX environment variable and where I can access it to change it? Also where I can find information about it? Regards Bilal Ghayad