similar to: Testing the manager.conf: sending and receiving commands

Hello all Has anyone had any success with the Manager API ? I am trying to check an extension status without too much luck I have the following <?php $fp = fsockopen("127.0.0.1", 5038, $errno, $errstr, 30); if (!$fp) { echo "$errstr ($errno)<br />\n"; } else { $out = "Action: Login\r\n"; $out .=

Hi All; I am configuring IAX endpoint, I just need to understand why I have to set requirecalltoken = no to be able to register because the following message is displayed: [Mar 22 12:25:39] ERROR[2297]: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 78.154.240.146 in the calltokenignore list or setting user gwbilalkwpciax

Hi; If my landline service provider does not provide the ability to send the SMS, and I need to send SMS from asterisk, then what is the required? How? Is it possible to send SMS from asterisk using SIM card to be connected via GSM adaptor connected to FXS ports? Or HOW?

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

OK Dears; Is the exten => _X.,2,Wait,2 no more working with Asterisk 1.8? What is the equivalent? I installed Asterisk 1.8 and Star2Billing 1.9 but I am getting this error if someone can advise me: Executing [9615806234 at a2billing:1] Answer("SIP/gwsshihabuddinkw-00000014", "") in new stack [Apr 5 02:59:05] WARNING[2941]: pbx.c:4055 pbx_extension_helper: No application

Through the AMI how can I tell if a call is on hold or not? I am using 1.4.X Thanks, jerry

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 All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal

Hi Danny; Really I did not understand how I can determine if the IO will be DB or conf files? Is it from the Asterisk manager? Regards Bilal ------------------------- It depends on how you are configured. The gui interfaces using Asterisk Manager, so you get the Same IO from the gui that you would get from a native manager session. -----Original Message----- From: asterisk-users-bounces at

Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: "It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the

Hi All; Anyone knows if there is an Asterisk version that setup can be stored in Database instead of the configuration files (.conf)? Any advise? Regards Bilal

Hi List; How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed

Hi All; Till now I am not able to find a good IAX IP Phone or Gateway that can be used with good quality. Anyone can advise for good one? Regards Bilal ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost.

Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards

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

Hi All; I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8? For example, when to decide that I have to go for 1.6 or I have to go for 1.8? Regards Bilal

Hi All; I tried Firefly softphone with IAX and it gave very poor quality. Any one advise a good strong softphone that can work with IAX fine? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

Hi All; Any one can suggest a TFTP server to be installed in Fedora (same machine that Asterisk is installed) to be used for Cisco IP Phones to download the required firmware and configuration files. Thanks for the help in advance. Regards Bilal

Hi All; My ISDN was working fine, and suddenly I start getting the below: sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel anyway! There is a Yellow Alarm, so what it could be the problem? My configuration as following: system.conf span=1,1,0,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 chan_dahi.conf context=IncomingPSTN group=0 signalling=pri_cpe switchtype=euroisdn

Hi List; What is the difference between WaitExten function and TIMEOUT (response)? As I see that both are used to determine the allowed time to enter the digits, any one can advise? Regards Bilal ____________________________________________________________________________________ Shape Yahoo! in your own image. Join our Network Research Panel today!