similar to: asterisk nagios plugin

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:

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:

Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. I know I can porobably go through the CDR logs and look for calls which have overlapped in time, but I'm wondering if there's some counter somewhere I could access... (I'm looking for evidence for an ISDN client who

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>

I am having some issues when trying to connect with perl to the asterisk manager and doing an "IAX2 show channels". If i do that on a server that is heavily loaded, i sometimes get some events instead of the channels i asked for. Any suggestions how i could fix that ? zoa.

I know that I can record the contents of a call by calling Monitor() or MixMonitor() from the dialplan just before invoking Dial(). I have a potential customer who wants only the first minute of each call recorded (for identification purposes, without the storage overhead of keeping the complete call). Can anyone here think of the easiest way to do this? The only possibilities I can think of

Dear members of the list, I have difficulties to obtain sync with a Digium TE420 PCI Express For four entries E1, In my case I am with only 3 E1s available to configure. The telephony operator is BrasilTelecom and Signaling is R2 Digital. The information I have is that are E1s with 32 channels, not have sure. My settings Have been based on the link reference:

I keep getting a red alarm when trying to setup asterisk to use my TE420B EC. I only have a blank context setup in my extensions.conf as I haven't started to config that until I can clear this red alarm. I don't have physical access to the server, so I can't go reseat the modules/card/ethernet cable, though I have hands on location that have done this a couple times

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello List, since some days i run into the problem that one span on a TE407P is not comming up correctly. With intense debug on that span i get: < [ 02 01 7f ] < Unnumbered frame: < SAPI: 00 C/R: 1 EA: 0 < TEI: 000 EA: 1 < M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] < 0 bytes of data

Dear, I always had one E1 card with one span, so I've never had any problem in set it up through /etc/dahdi/sustem.conf and /etc/asterisk/chan_dahdi.conf because I put span=1. But now I have a PBX with two E1 cards with 4 span (8 span in total). How do I have to define both card in system.conf and chan_dahdi.conf, and how do I have to refer each span to the corresponding card ??? Thanks a

Hi anyone have a idea for debug my digium TE405P card ? My zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = fr defaultzone = fr My Zapata.conf: [channels] language=fr context=from-E1 switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

hi folks. recently some of our customers complained about bad voice quality on the phone system. i looked at the logs and found a lot of these: [2012-11-03 08:26:38] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:45] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:54] NOTICE[11305]

I have a box with 12 T1s (4 Te410P cards). The PSTN provider is reporting slips and ask me to update the clock source. I have my system.conf set as the following but when I run dahdi_scan only the ports on Card 1 are showing up with syncsrc=1 system.conf : span=1,1,0,esf,b8zs bchan=2-24 mtp2=1 span=2,2,0,esf,b8zs bchan=26-48 mtp2=25 span=3,3,0,esf,b8zs bchan=49-72

I just managed to put in a TE410 card in an Asterisk box to work with OnRamp 20(E1 downunder). I am able to dial in but was not able to dial out. Can anyone offer me some advice please? In my extensions.conf, I just put in: [default] ... exten => 0,1,Dial(Zap/g1) and I get this on the console when I dialled 0. -- Executing [0 at default:1] Dial("SIP/5166-b76004f8",

We are using AAH with Asterisk 1.2.7.1 with a TE405P as listed below. We are getting frequent restarts on the spans which lead to dropped calls. I have pasted some hopefully pertinent information below -- anyone have any clues that might help? Thanks Next line is repeated throughout messages, going through every channel in every connected span. asterisk/full.1:Apr 24 01:15:25 VERBOSE[4196]

Hi, after loading the zaptel driver wct4xxp I have strange log lines in the syslog: Out of storage space. free tells, that more than 900 MB are still free. Disk space is also available. I'm using a dual opteron in 64 bit mode. Any ideas? Roger. Syslog: Dec 1 02:18:37 ipphone4 kernel: TE410P: Launching card: 0 Dec 1 02:18:37 ipphone4 kernel: TE410P: Setting up global serial

Hi, I'm really lost on the directory functionality of the asterisk. I am currently setting a ACD using asteriskathome. I want to use the # feature where in caller can access the companies directory. My issue is how to setup the entry on the directory. Sorry for my silly question Can somebody guide me please. Regards, Leonimar Cape __________________________________________________ Do You

Hi All, I am facing problem with E1 line. I have installed Asterisk (1.4.20.1) on a system with Digium TE420 card (Zaptel- 1.4.10) But every now and then I face problem of down E1's. The log show lot of entries like "pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2" This happens on a regular basis and the E1 becomes up after some time. My

Hi, I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card 5.0V (rev 02)) 4 ports I want to make a loop test between digium card E1 to test the configuration of dahdi What I want to do scenario is I connect port 1 and port4 in the digium card with E1 cable SIPcall-->E1 Digium port 1--->(Loop)E1 port 2---->sip extension local. kindly can any can help me to

List: This is my first manager client that I've written so please bear with me: I am trying to write a C manager interface client to interface with our CRM software. I am having an issue while reading the data from the manager interface. I am writing this in C and I have the following code: while(1) { bzero(buffer,sizeof(buffer)); readCode = read(socketHandle,buffer,sizeof(buffer));