similar to: Which IP Phones have buttons can be assigned to functions with Asterisk

Hi List; How to upgrade the Asterisk, Zaptel and LibPri and keep the configuration the same? I do not need to remove current asterisk, zaptel and libpri and download new one and write new configuration. Regards, -------------- ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460

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; Did any one tried the H.323 module? How much it is stable and work fine? Regards, ------------ ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 ____________________________________________________________________________________Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/

Hi List; How can I configure asterisk to receive a call from SIP end point without being registered at asterisk and its IP address is dynamic, and authentication to be based on the username and password or any other string? I know that if I place the host with static IP then no need to register, but what if the voip gateway was having dynamic IP and I do not need to register on asterisk, but I

Hi List; All we know that in voice, there are a type of communications between endpoints, for example: in some communications we do a proxy for media and signaling while other communications we do a proxy for only signaling. Where I can determine these things in Asterisk if I am using SIP and if I am using H.323? Regards -------------- IP Telephony and Contact Center Engineer Eng. Bilal Ghayad

Dear Alex; Thanks for your kindly reply. Please explain for me what do u mean exactly in "a la" in the following sentence u wrote it below? " in SIP, this can be done via "re-INVITEs" a la the canreinvite= option for SIP peers in sip.conf" Another thing, do u mean that it is easier (better) if we need H.323 endpoint to talk with SIP endpoint then we use full

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; What is the command and where I can write it to block specific code from calls (then no one will be able to place call for any number start by that code)? --------------- Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: + (965) 9849460 Yahoo ID: bilmar_gh at yahoo.com MSN ID: bghayad at hotmail.com

Hi List; In the extensions.conf file, at the [global] context, there is a variable configured as: CONSOLE=Console/dsp What does it mean that? What dsp mean and it is shortcut for what? How can I use the core to get some data about such thing ambiguous for me? Regards, ---------- Bilal Ghayad ____________________________________________________________________________________ Be a

Dear List; Thanks alot for the help. But how can I let the second dial tone (after pressing the extension to select that FXO port) to be difference than normal dial tone? Regards Bilal Ghayad -------------------------- Correction, on FXO port not FXS, second, read his email first: "Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP

Hi List; I compiled Zaptel 1.4 and Asterisk 1.4 after downloading them using svn, but when I checked the file zaptel.conf under etc/asterisk, I did not find this file. Any help? By the way: How can I know the asterisk and zaptel version extactly that I compiled them? In other words, asterisk 1.4.... and zaptel 1.4.... ? Regards ------------- ITS IP Telephony and Contact Center Engineer Eng.

Hi list; ASTCC supports IVR or there is a separate module for IVR? Can someone advise me a link to start download and ready about ASTCC to do the configuration? Regards, --------- ITS IP Telephony and Contact Center Engineer Bilal Ghayad Mobile: 00865 9849460 ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play

Hi List; Can asterisk hear (receive) calls on two IP addresses? How? If yes, then: If I have a VPN router, and my Asterisk server connected to two network cards, one has a private IP address (192.168.0.2) connected to the VPN router (192.168.0.1) and another network card has a private IP address (193.111.196.249) connected directly to the outside default gateway (193.111.196.240), where the VPN

Hey everyone, I know this isn't a direct Asterisk issue, but some of you may know this answer. I recently upgraded the SIP version to 1.6.6 on all of our phones in the office. Everything is working fine, except one aspect. The phones in the office reboot randomly for no apparent reason. I haven't changed anything in the configuration files since the upgrade. The only setting in the

Hi, At one location we have a user whose Polycom IP430 suffers from erratic reboots. We swapped his phone for a brand new one, but the problem remains. Has anyone seen that?

I believe the CorePlayer will play back FLAC. Atamido ----- Original Message ----- From: "Josh Coalson" <xflac@yahoo.com> To: "Harry Sack" <tranzedude@gmail.com>; <flac-dev@xiph.org> Sent: Thursday, March 22, 2007 6:26 PM Subject: Re: [Flac-dev] FLAC: players for Pocket PC > http://www.google.com/search?q=flac+pocket+pc > > e.g. > >

If I use a call file (/var/spool/asterisk/outgoing) is it possible to have a video phone connect to a camera (linksys wvc200) and show the camera stream and hear the audio? How would I do that? Jerry

Hello, I follow the instruction of this link (http://wiki.dovecot.org/LDA) to implement sieve with my dovecot installation (dovecot-1.0rc2) and i'm running the timsieve daemon of my cyrus-imapd installation but i got the following error message when the sieve daemon try to formward message : entered bc_action_emit with filelen: 16 Aug 13 14:53:27 ldap sieve[3870]: entered bc_action_emit with

Does anyone have a script that will notice a Rumplestiltskin type spam attack (where they try every name possible) and drop the sending into a block list? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla@yahoo.com -- This message has been scanned for viruses and

Hello all, cell <-T1-> zap <-internet-very remote-> sip (ip430) The audio is choppy ONLY to cell USER. The polycom user says the audio is fine. SIP-SIP calls sound good for both parties. Where should I setup the jitterbuffer? The zapata.conf (recent * 1.2) and/or the polycom configs (fw 2.0.3)? Any tips with the zap or polycom settings below would rock. Packet loss - average