similar to: SIP call hangup from asterisk CLI

My Inbound calls lands , but line get disconnect in exactly 3 secs. Here goes my extension.conf setting : [from-ipkall] exten => 901835,1,Ringing ; call ringing exten => 901835,2,Wait(1) ; Wait 1 second for CID delivery from PRI exten => 901835,3,Answer ; Answer the line exten =>

Hello; I'm running a FreeBSD 4.10-release-p2 box and both chkrootkit 0.44 & 0.45 report that my /sbin/init file is infected. It appears as though the egrep for "UPX" in the output of "strings" triggers the infected notice. When I copy the init file from an uninfected box to this one chkrootkit continues to report it as infected. Is chkrootkit reading a copy of the

Hi, I'm trying to call a SIP server while providing the SIP server username/password in dial string but it's not working ... Dial(SIP/gs102:test at 192.168.2.81); User on sip server (192.168.2.81): [gs102] disallow=all allow=ulaw allow=alaw type=friend username=gs102 secret=test host=dynamic dtmfmode=inband defaultip=192.168.2.1 qualify=1000 mailbox=102 context=context-gs102

We are using G711 for phones to talk to Asterisk and G729 licenses at asterisk to talk to ITSP Could you please suggest transcoder to use from G711 and G729 and which is comptible with Asterisk. We will like to avoid using TDM if possible Also i remember that initially we didn't have G729 and were using only 711 for with vicidial but then also we had same problems. at that time it was only 2

Hi. I have newely installed vicidial now i am getting thise error anyone can hel me. NOTICE[31819]: channel.c:1972 ast_read: Dropping incompatible voice frame on Local/8600051 at default-ed7b,1 of format gsm since our native format has changed to slin Regard's Vijay Kumar -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list. I have Asterisk installed on a Debian 1.8 6 64-bit. What happens is the following, some channels are not being hangup properly. They run the hangup in dialplan, but the output of the command "core show channels" shows several channels with status "rsrvd." Checking the server's memory, the "top" command shows multiple processes and stopped using the

I have two Adtran 750's connecting our analog phones to asterisk. On occasion, I get a channel that gets "stuck" off hook. 'show channels' shows: Zap/27-1 (longdistance s 1 ) Rsrvd (None) (None) And will just stay like that until the phone is manually picked up and hung up again (or asterisk is stopped/started). I guess this is a function of an unclean hangup (being

Hello, I've installed VICIDIAL per the instructions on the astGUIclient website. It appears everything is working correctly. All the conference rooms have been set up, the database is running, and all the astGUIclient/VICIDIAL scripts are running. I'm using the VICIDIAL client on windows 2000, and it also appears to be working correctly. I can log in with no problems with the user

Hi When i ran the below command on vicidial dialer: [root@vicidial2 ~]# tethereal -i eth0 -a duration:300 -w sample.cap Capturing on eth0 320167 147496 packets dropped on net i found: When i ran Acterna PVA-1000 on sample.cap it showed Max Jitter about 20 % and packet loss and echo as major cause of voice degradation. MQS was also less 2.59 where as it should be around 5.0. are packets being

I have a small system based on one TDM400 card with - 2 FXO modules for incoming lines - 1 FXS module for one phone The system was working fine in the past. The motherboard was exchanged and during the switch, the phones line was rewired with a mistake and a incoming phone line was connected to the FXS module, and there was ringing voltage on the module. Now, the system kinda works but the FXS

Hi, If I understand the MS documentation right a Windows desktop uses DNS (or WINS) to determine a DC and then connects to the ldap udp port 389 to get further details about the DC. I do not want to run a full DC but only a "simple" Heimdal or MIT kdc for Browser Negotiate authentication. Does Samba include such a standalone ldap service which just returns the only supported

hello, I got those extensions: exten => 555,1,MeetMeCount(500|count) exten => 555,2,Gotoif,$[${count} = 1]?6 exten => 555,3,Meetme,500|pMs|1234 exten => 555,4,Playback,goodbye exten => 555,5,Hangup exten => 555,6,Goto(from-internal-custom,556,1) exten => 555,7,hangup exten => 556,1,System(/bin/cp /etc/asterisk/1-test /var/spool/asterisk/outgoing/) exten =>

I'm running Asterisk CVS-v1-0-12/20/04. I'm using PHP with Manager API Here is the code: #################################################################### # Make call #################################################################### $socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout); if (!$socket) { echo "$errstr ($errno)<br /\n"; } else {

Hello I need to write a script that will dial a list of customers and play a message. I couldn't find a way to tell Asterisk/Zaptel to wait until the callee has actually picked up the phone before proceeding with Playback(): ============ ;call made through Dial(): Doesn't proceed after off-hook/hangup [internal] exten => 8888,1,Dial(Zap/1/${IPPI}) exten => 8888,n,NoOp(We never

Here is another log from the * server CLI, I reall hope some one can help me out on this one. thanks |SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0.0.1' and campaign_id = '' and call_time < "" and lead_id != '';| -- VDAD get agent: |0|update of vla table: |127.0.0.1 |UPDATE vicidial_live_agents set

Hi All. I have many days reading and research about asterisk and vicidial. I thing this issue is about asterisk and doesnt about vicidial. Isn't it? I have a problem with theses application (I already ask for help in vicidial forums), but I can not fix it. I have debian 5 with asterisk 1.2.24 and vicidial 2.0.4. This server has a IAX tunnel with another asterisk server B which connect to

Can anyone point me to a good howto or example on how to get * to recognize inbound faxes and adjust accordingly? Ideally I would like it to grab the fax and email it to me, but I dont know if that is really possible yet or not. Thanks Neri -------------- next part -------------- An HTML attachment was scrubbed... URL:

My system is linked to a legacy PBX via Q-SIG and most of my tests so far have been from that PBX. I created a number of MeetMe conference rooms and they work fine when called from the legacy PBX. However, when there's a MeetMe room with a legacy caller and a SIP phone, the SIP phone can hear the legacy caller. But the legacy caller can't hear the SIP phone. However, "meetme show

Hi With an FXO module + Zaptel, I'd like to know if there are ways to know when the remote party has answered the phone, whether calling through a callfile or by sending DTMF's. I read about {CHANNEL(state), ChanIsAvail(), and ${DIALSTATUS}: Are those reliable ways to know when the channel is available for dialing out and the call has been answered?

Vicidial can't call and transfer to my softphone. I get some line that says Spawn Extension....exited on non zero.... Here's some of the CLI output. I am using Asterisk 1.2.4 and astguiclient 1.1.8 ...thanks for the help |SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0.0.1' and campaign_id = '' and call_time <