search for: writeprotection

Is there anyway to unset the extensions.conf definition of writeprotect=yes while in the CLI interface (or by other mechanism) to enable the dialplan save command? I accidentally overwrote my extensions.conf but still have a running copy of asterisk with the old dial plan running in memory. while it would not be difficult for me to rebuild what I lost - it would be easier if I could just save

Ok. I am confused. I now have conflicting answers to my question: Do you need to use a special phone to use SIP? My setup is X100P and TDM10B. I would like to connect to iConnectHere, which uses SIP. Has anybody done this before (using similar equipment to what I have listed above)? And if it is not possible, could somebody please explain why. I don't understand why this wouldn't

Hello all, I'm running Asterisk 1.4.5 and Zaptel 1.4.3 on Debian Etch i386 with the Digium's Dev Kit that comes with 1 FXO and 1 FXS. How do I configure my home PBX in such a way that whenever someone calls on my trunkline (PSTN) number, he/she will hear a customized ring tone, probably playing an MP3 file, instead of a boring standard ring tone while the extension number that is

in sip.conf [general] port = 5060 ; The TCP/IP port for SIP communiations bindaddr = 0.0.0.0 ; Address to bind to. 0.0.0.0 all addresses on server. context=other ; Default for incoming calls disallow=all allow=ulaw allow=gsm in extensions.conf [general] static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here

Hi I have reason to believe that I have errors in my configuration because when I make a call I can see the H323 call executed ok but not being processed by Zap. I am using R2 signaling ( which I know is incomplete but should I not see it when I debug Zap channel?). I think there is a problem with my Zapata and zaptel configs . I understand that R2 can work with R2 China and R2 Argentina.

I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) answers an incoming call and forwards that call to a SIP softphone (X-lite.) Seems all is built/installed okay: # ztcfg -vv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. I'm pretty new at this and the extensions.conf file is eating my

Hello, I'm just starting out with asterisk and I'm playing around with the system. Currently I have a Digium TE210P connected to a PRI on the Asterisk server. I have a SIP soft phone on my laptop for testing that is working fine. When I try to place a call from my soft phone I get this from Asterisk: May 1 09:11:41 NOTICE[20098]: app_dial.c:1029 dial_exec_full: Unable to create

I would like to create a simple menu that would allow a caller to decide whether they want to leave a message or be forwarded to another number (i.e cell phone). Thanks in advance for any insight. Here's my current extension.conf [general] static=yes writeprotect=yes [globals] [default] exten => 101,1,Dial(SIP/101,20) exten => 101,n,Voicemail(101 at default) ;This automatically

Hi! I have the chance to play with a couple of E400P cards, each installed in a IBM e330 XSeries servers (2 x 1GHz P-III CPU 2 Gb RAM, 36Gb SCSI HDD with RH8.0 2.4.18-smp kernel), and I'm trying to test/benchmark this e330/E400P combo generating calls thru /var/spool/asterisk/outgoing One e400P if doing the carrier work making calls and the other just receives the calls: Server#1

hello, I have a test setup with 2 asterisk servers, each having a one snom 100 via sip using it. I`m experimenting on how trunking between them would work. I have them setup for RSA authentication which I plan to use in the future. So I`ve setup the keys and servers seem authenticate to each other. One is named phila and other hurricane. Here is what I see on phila: -- Registered

Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk

I have this in my dialplan... [general] static=yes writeprotect=no clearglobalvars=no [start] exten => 5000,1,Answer exten => 5000,n,Wait(1) exten => 5000,n,NoOp(${CALLERID(num)}) exten => 5000,n,Playback(tt-monkeys) which, when I dial 5000, executes this... == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [5000 at start:1]

Hello list, I've posted my problem on BSD list and i still have the problem. The remote side receives the call , but there's no voice on the call. I tried everything about possible NAT problems .. but ther're on same net. My platform: FreeBSD 5.2.1-Release Asterisk 1.0-RC2 soft phones : X-Lite >>>> -- Executing Dial("SIP/1260-a7ae", "SIP/1262|20")

Hi, I have the Cisco PIX 506 firewall right in front of the asterisk and I am getting a one-way audio. I need your help/guidance to resolve this problem. I have the "fixups" disabled for SIP in the Cisco PIX 506. Any help rendered by you in this subject is greatly appreciated. I have been breaking my head trying to resolve this problem for more than one month. I have included the

Hi, I am trying to forward calls to another * server with IAX Here is What I want to Do 1- Call SERVER1, let say at 51412345678 2- SERVER1 should transfer the call to SERVER2 in a remote location 3- SERVER2 Receive the call and transfer it to the PSTN number. I have one X100P card on each machine. What is happening is that when the remote party picks up the phone, all he can hear is a weird

Hola! I'm using asterisk as H.323 -> PRI gateway. First call goes thru ok, second concurrent call fails with: Aug 6 11:52:30 DEBUG[737292]: chan_h323.c:1038 setup_incoming_call: Sending to context [ip2pri] -- Executing Dial("H323/ip$192.168.32.25:60271/984", "Zap/1/9541163107100") in new stack Aug 6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to

Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no

Hi, I just installed OH323 Plugin and im now tryin to make simple Configuration to connect Openphone and Xlite to my Asterisk-Server. All works fine, i just wanna know if there's a better way to do it? Is there anything wrong with my Config? OH323.conf [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=8000 udpEnd=8005 fastStart=no

Hi, Could someone help me on configuring a H.323 trunk. I am trying to set up the following scenario: [SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)] I am using the following versions: Linux CentOS 3.3/2.4.21-.EL.co asterisk 1.0.1 pwlib_1.5.2 openh323_1.12.2 asterisk-oh323-0.6.3b Calling from Asterisk (2004) to the

Dear all, I've been taking my baby-steps toward setting up an Asterisk phone system in my office, as also between my home and office (connected by DSL). I'm have a rough time getting two * boxes talk IAX over a LAN. I don't know what I am doing wrong, but am attaching my iax.conf and extensions.conf on both the boxes. Does anyone see it? ------config files start------ site-0