similar to: Asterisk GUI

Hello friends, I am getting this error:- Oct 23 15:47:22 WARNING[2124]: db.c:171 ast_db_put: Unable to put value '192.168.1.12:5060:300:15553695861:sip:23@192.168.1.12:5060' for key '23' in family 'SIP/Registry I have no idea what it means. Please tell me what could be the problem. With warm regards. Vivek J. Joshi. vivek@staff.ownmail.com Trikon electronics Pvt. Ltd.

Hello friends, Can I run two asterisks running simultaneously on the same machine? I want one to run v1.0.2 for h323 ( which is an old and running production system ) and one for sip implementation. I wonder how it can be done since they will want access to the same ports and ip addresses. Does anyone know to do this or has done this before? Please share your experiences please.

Hello friends, I am a newbie to asterisk , please help. I am receiving a phone from a sip server and I want to route it to another sip server. The problem is that the target sip server takes a # in the argument . I am trying to dial as exten => s,5,Dial(12345#123456789@xxx.xxx.xxx.xxx) ;;;; dial the number at the ip address. When this gets executed, I get the following eror:-

I have an ATA-186 in a SIP configuration (following Shawn Djernes how-to), but I get the following error at the asterisk console when I try to call the phone connected to the ATA: ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Everything works if I remove indications.conf from /etc/asterisk -

hi all im trying to compile h323 i have got the pwlib and openh323 working that is simph323 is running properly but when i try to compile h323 in the channels directory it gives me the following error can anybody please help me with [root@test src]# cd /usr/src/asterisk/channels/h323/ [root@test h323]# make opt g++ -DNDEBUG -I../../include -Wmissing-prototypes -fPIC -DP_LINUX=2.6.5-1.358

Hello friends, I am trying to install asterisk 1.4.0 . I am configuring it as follows:- ./configure --prefix=/home/vivek/downloads/install/asterisk/ But still while running 'make install', it tries to install it in /var/lib/asterisk/ and stops because of failing permissions. I have provided it a prefix, But it doesn't install it there. Can anybody tell me the solution for this. I

I have an ISDN line with 10 numbers. The line is then connected to * with one HFC-based card. The format of the numbers is like below: 123456-0 123456-1 ... 123456-9 Now I would like to connect those numbers to different telephones, i.e. when someone dials 123456-0, he/she is connected to the digital receptionist. If someone dials 123456-2, the connection goes to SIP/202 If someone dials

an obvious fix to PIC IO intercept. In PIC IO, address from send_pio_req is physical address already. Signed-off-by: Xiaohui Xin <xiaohui.xin@intel.com> Signed-off-by: Xiaowei Yang <xiaowei.yang@intel.com> Signed-off-by: Xin Li <xin.b.li@intel.com> _______________________________________________ Xen-devel mailing list Xen-devel@lists.xensource.com

Hi, I am trying to get the extension status (weather it has dialed outgoing call via SIP or IAX2), using the following piece of code however it always returns -1 on all the extensions (valid/invalid). Am i missing something ? Any help. Thanks ----------------------------------- #!/usr/bin/perl use Asterisk::Manager; use lib './lib', '../lib'; $|++; my $astman = new

Hi all: I have a question about lm on t-test. data(sleep) I wanna perform t-test to test the difference between the 2 groups: I can use: t.test(extra~group) The t.test result shows that:t = -1.8608; mean1=0.75,mean2=2.33 But I still wanna use: summary(lm(extra~group)) Intercept=0.75,which is mean1,just the same as t.test. group2=1.58 means the difference of the 2 groups,so

Hi, I'm getting an issue while executing AMI Originate. I'm getting "extension does not exists" on Originate's Response, and on the other hand Asterisk CLI say "fwrite() returned error: Broken pipe" Please suggest me what is wrong. Muhammad Faheem ### my originate code block ...

hi all i'm getting the below error when trying to compile asterisk-1.4 on redhat-9.0 any suggestions ? make[2]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' make[1]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' Generating input for menuselect ... menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... [CC]

hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only

Hello , I have a problem to install asterisk with ldap. I am doing the following: make clean . / configure make menuselect LIBS =- lldap export LIBS make ====> This is where my error #make CC = "cc" CXX = "g + +" LD = "" AR = "" RANLIB = "" CFLAGS = "" make-C menuselect CONFIGURE_SILENT = "- silent" menuselect make [1]:

Hi, Does anyone have some experience with Nokia 32 Terminal (it is an analog GSM Gateway)? After a configuration I can make only incoming calls, I'm not able to do any outgoing. Nokia signalize an error (4 short tones), when I try to phone someone. I tried postpaid simcards as well as prepaid simcards with the same result. Does anyone try to connect this gateway to Asterisk PBX if so what

Has anyone used Gastman/Astman successfully? I have it up and running (Gastman win32), but have a problem with the creation of end stations on the map. I'm not sure of the format of the extension to use when creating a end station icon. Services like Conference bridge and Musichonhold seem to work ok (I use 555@mainmenu and 666@mainmenu) for the Icon extensions. IAX softphone seems to work

Dear All, I have a requirement to 'originate' a number of calls to various external users from within a conference room, so that the end users does not pay for the call. I know that within Astman I can define an extension and then originate the call from that extension. Can I define a conference room (how would I configure that on astman? What channel would it use?) and then generate a

I have been trying to draw histogram for my manscript and found some strange things that I could not figure out why. Using the same code listed below I have successfully draw histograms for a few figures with fraction labeled on Y axis less than 1 (acturally between 0 to 0.1). But one dataset gives the Y axis label 0 to 5 as fraction. This is not true, as fraction are less than 1, although the

Hello All, My build fails in the "make asterisk" section, here is the log of the final steps it performed. All help appreciated... Thanks and Regards Ajit ============================================ make[1]: Leaving directory `/usr/src/asterisk/cdr' make[1]: Entering directory `/usr/src/asterisk/astman' rm -f *.o astman make[1]: Leaving directory

hello i am getting this error while trying to run ast-rad-acc.pl my $ast_connected = 1; while( 1 ) { if( $astman->connect ) { $ast_connected = 1; syslog('info', 'Connected to Asterisk!'); $astman->setcallback('DEFAULT', \&status_callback); eval { $astman->eventloop; }; print STDERR "Connected To Asterisk\n"; } else {