similar to: Asterisk H.323 channel...

At time to time I receive some junk faxes from some advertising companies that play smart and don't provide any TSI number so I can not bock them by the number in Hylafax. Despite calling their Fax Removal Service 1-800-... number several time they refuse to obey my request. So I would like to setup a small script or context loop in extension.conf if possible and maybe run it overnight; maybe

Hi, I created a script named example2.sh which goal is read some text from my HP Service Desk using an application in java and send this text to the text2wave application for TTS. example2.sh java -Xbatch Example10 | text2wave -f 8000 -o /var/lib/asterisk/sounds/my-sd.wav When I execute the script in prompt, everything is ok, but when I use the system() command in my extensions.conf it isn?t

I'm trying to capture faxes on 1.6.1-beta4. AFAICT, app_fax is working OK. I'm then using fax2mail to send the fax. That wasn't working, so i posted for help using the System() cmd, since fax2mail did work from the command line. But now I realize it's fax2mail and mime-construct itself. I set up a fax-test context: [fax-test] exten=>666,1,NoOp( fax-test )

Does anyone know how to get CID working in AEL2 ? In extensions.conf you can do: exten => 111/666,1,PlayBack(demo-congrats) exten => 111/666,2,Hangup() exten => 111,1,PlayBack(demo-moreinfo) exten => 111,2,Hangup() and if callerid 666 dialed 111, they would get demo-congrats, everyone else gets demo-moreinfo. In AEL: 111 => { Playback(demo-moreinfo);

Hi, I have a bunch of SIP phones(behind NAT) registering on my * box. I want to find out when they register and de-register. I also want to operate on it, so when they register/de-register, I want to insert calldate into a mysql DB, etc..... Maybe this will help me when, for instance a user tries to register but has the wrong username/password. Now I am aware of regcontext, but it only

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

Hi all, I am trying to get a Menu system to work, and having probs with the internal extensions from the prompts. Below is the extensions.conf section. [mainmenu] ; ;"main menu" context with submenu ; include => default exten => s,1,Answer exten => s,2,Background(hello) exten => s,3,Background(thank_you) exten => s,4,Background(if_you_know_extension) exten =>

Dear All, I bought one IP PHONE from Integrated Networks which was showed to wiki too: http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks I have problems with the Asterisk authentication. It does't want to LOG IN to Asterisk; it always says "LOG ON FAILED". I'm using the IAX2 protocol and all paramters seems to be correct. Does somebody use the same IP PHONE with

Hello People, I'm newbie in * 1.0.1, running a Linux 2.6.7 in a Debian Sarge, and this is my situation: +------------+ +-------------+ | Sip Server |-------------|CISCO PSTN GW| +------------+ +-------------+ \ || \ || \ +----------+ || | Asterisk |=========

I have a DID from IPKall.com which is forwarded to my asterisk box. Then this extension should call my ip phone using Dial application. Everything works fine, except when I pickup the phone, I can talk, the other party can hear me, but I cannot hear anything the person says on the ip phone. Then after a couple of seconds, the call hangs up. I don't know why. Here is the message I get:

Hello, I'm configured Sipura-3000 to forward IP calls to PSTN number on no answer (In User1 tab Cfwd No Ans Dest: 123456@gw0) IPPhone ---IP---> Sipura-3000 ---PSTN---> PSTN User Generally it works fine, but Sipura sends back SIP OK to IPPhone just prior to dialing to PSTN number. How to configure Sipura to detect that the remote side on PSTN picks up the phone and only then to

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

Dear all, My office have an old asterisk PBX system (asterisk 11.4), and it encrypt all the SIP User's secret. But the voip engineer before me didn't save / documented those password. Now the server's hardware is begin to broke, it hangs a lot, and have a lot of call problem. We already have a new asterisk PBX to replace it, but we have difficulty to retrieve the encrypted password.

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

I have recently found out that * is very strict about dialing out. If a number isn't listed in extensions.conf, good luck trying to dial it. I had to put in a line for each of our area codes with XXX's before I could dial local numbers. Anyway..now that I 'can' dial them, as soon as the other party picks up the phone I get a busy signal on my end. Also..just tried an IpPhone to

Hello again :-( I have a problem with chan_unicall. If I have two simultaneous incoming or outgoing calls, they sound broken because cpu load goes to 99%. Also with one call, the cpu load goes to 99%. Seems like device /dev/zap/channel is busy after 5 or 10 seconds , and chan_unicall does not write to this. strace with asterisk-1.0.7, zaptel-1.0.7, kernel-2.6.10 ================================

Hi All, I am getting this error when i try to compile the "Linphone" package by typing----- make. please help me i am feeling very frustrated with this error pasdt from 7 days i am getting this error. please help me. speexec.c: In function `speex_ec_process': speexec.c:112: `spx_int32_t' undeclared (first use in this function) speexec.c:112: (Each undeclared identifier is

Hello, I've one astribank with 8 FXO unit and 8 pstn lines connected to the astribank. When I receive calls on my ipphone I get always Unknown callerid. It's is possible to receive the callerid from the lines on the astribank unit? This is my config: [channels] language=es context=from-zaptel signalling=fxs_ks ;rxwink=300 usecallerid=yes callerid=asreceived ;cidsignalling=bell

Hi Guys, i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way: 1) use a phone in PBX1 2) call extension in PBX2 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a cellphone) my questions now is : am i gonna be able to dial from an IPphone registered within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2? anybody know