similar to: asterisk manager and perl

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ CLI Output : ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vicidialnow*CLI> == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from

Hi all R-helpers, i am a new R-user and have problem with deleting some components in a column. I have a dataset like Name Id x empty 2 empty 3 a none 2 b none 3 d none 2 ad cfh 4 bf cdt 5 empty 2 empty 2 gf cdh 4 d none 5 and want to

I am trying to setup Asterisk as the voicemail server for Cisco Call Manager. I have just about everything working except for the message waiting indicator. I have the following setup in context [ccm] in my extensions.conf file: ;MWI exten => _2807XXX,1,SetCallerID(${EXTEN:3}) exten => _2807XXX,2,Dial(SIP/28888@65.202.115.240) exten => _2807XXX,3,Answer exten => _2807XXX,4,Wait,1

Hi, I want to pick up a call with the snom's programmable buttons(snom190 -SIP 3.60x, snom360-SIP 4.1) with asterisk server (v 1.2.0), I tried with the option 'Destination' and when the incoming call arrive to another snom phone the button blinking. In this way I can only pick down it pressing the blinking button. The solution is call the *8 or parcking the call but my pbroblem

I have a setup with lot's of small files (Maildir), in 4 different volumes and for some reason the volumes are full when they reach 60% usage (as reported by df ). This was ofcourse a bit of a supprise for me .. lots of failed writes, bounced messages and very angry customers. Has anybody on this list seen this before (not the angry customers ;-) ? Regards, =paulv # echo "ls

I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed the WIKI page on setting it up but I can't seem to get it to work. Here is my Asterisk version: pbx1*CLI> core show version Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on 2008-01-10 12:08:48 UTC Here is a log of when the FollowMe is being called: NOTE: I've tried to use the AstDB as

Remove some pointless conditionals before kfree(). Signed-off-by: Wei Yongjun <yjwei at cn.fujitsu.com> --- fs/ocfs2/alloc.c | 3 +-- fs/ocfs2/cluster/heartbeat.c | 6 ++---- fs/ocfs2/cluster/tcp.c | 6 ++---- fs/ocfs2/dlm/dlmdomain.c | 3 +-- fs/ocfs2/dlm/dlmrecovery.c | 6 ++---- fs/ocfs2/extent_map.c | 3 +-- fs/ocfs2/journal.c

I'm not sure how much more simple I can make this but I just cannot seem to get my Asterisk 13 to accept a connection on the manager interface: --- manager.conf --- [general] enabled = yes port = 5038 bindaddr = 127.0.0.1 [myasterisk] secret=a permit=0.0.0.0/0.0.0.0 read = all write = all So, couldn't be any more wide open and simpler to connect yet: # echo -e "Action:

Hi Asterisk-users, I am working with Aterisk Manager API's. I can login successfuly with the following. char buff[256]; strcpy(buff, "Action: Login\r\nUsername: admin\r\nSecret: unix\r\n\r\n"); send(msock, buff, 255); Now I want to try Action: Originate, therefore I tried the following char buff1[256]; strcpy(buff1, "Action: Originate\r\nChannel:

Dear all, I want to make call through socket i have set code given below: #!/usr/bin/perl -w use IO::Socket::INET; sub asterisk_command () { # my $command=$_[0]; my $ami=IO::Socket::INET->new(PeerAddr=>'127.0.0.1',PeerPort=>5038,Proto=>'tcp') or die "failed to connect to AMI!"; print $ami "Action: Login\r\nUsername:

Hello, I have a question . dose asterisk supports "302 Redirecting..." ? I have SIP Server "Not Asterisk" and my Asterisk is registering as a client for this device . when i try to call another client registered to the same SIP server i got Busy Tone and here is the asterisk CLI output ----------------- -- Got SIP response 302 "Redirecting..." back

Hi, I have written a C code which would let me connect to the Asterisk Manager Interface. The code compiles successfully but on running the code I get unauthorized login shown in the Asterisk command line console. Here is my C code: #include<stdio.h> #include<netdb.h> #include<unistd.h> #include<string.h> #include<arpa/inet.h> #include<sys/types.h>

Hello all, I've got some PHP code that opens an AMI socket and does a ConfBridgeList for a specific bridge (8888). This all works just fine but I need to filter the information displayed to only CallerIDName so I can see a complete list of names of participants. After days of googling and playing with it, I'm no closer than I was when I started. I'm not at all married to a table.

Though I've had implementations of Asterisk, I havent encountered this one yet, so i'd like to seek your advise if this possible. I would want asterisk to be stand between the dialer the destination. The dialer will now dial asterisk access number. Asterisk will acknowledge user by using CallerID and check against its database for authentication and then sends out a DTMF A tone for ?

I have created a product, with two controllers and two models, one for categories and one for articles. In my models directory, I have article.rb and articles.rb and seem to have code that relies on both of these! I am unable to get any validation working because of this (whichever model I put the code in, it doesn''t load it) Should I just rip it all apart and start again? --

Hi folks, this does look like a bug to me: Asterisk replaces the @63.214.186.6 by @context which obviously leads to a failure. Any comments, do I have a configuration issue on my side that I missed? Cheers, Philipp -- Executing Dial("SIP/philipp-bd5f", "SIP/992365264680@nikotel- out|90") in new stack -- Called 99xxxxxxxxxx@nikotel-out -- Got SIP response 302

Greetings, I am trying to install R (0.63.2) on a Digital Unix 4.0b Alpha Server 2100A using gcc 2.8.1 and f77 v 0.5.2.3. It seems to compile OK. However, when I try to run R I get the following message: R : Copyright 1999, The R Development Core Team Version 0.63.2 (January 12, 1999) R is free software and comes with ABSOLUTELY NO WARRANTY. You are welcome to redistribute it under

Hello fellow asterisk people! I have Asterisk listening on port 5061 and SER on port 5060. Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP. My problems are with SIP. I can make incoming calls from SIP to asterisk and to any of the other networks, but when I try to make an outgoing call from Asterisk to SER I see the following in Asterisk: -- Executing

Hi to all. I'm trying to use meetme on asterisk 1.4.22.1. On a debian i've compiled (as i need h323 support) openh323_v1_18_0 pwlib_v1_10_0 dahdi-linux-2.1.0.3 dahdi-tools-2.1.0.2 asterisk-1.4.22.1 All works fine, dahdi status is: asterik:/data/programmi# /etc/init.d/dahdi status ### Span 1: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: RTC) 1" (MASTER) asterik:/data/programmi#