similar to: Time out not working from php agi...

Hello list. We just upgraded to 1.6.1.11. We are using real time information stored on mysql databases. That is all running fine. Now, since we upgraded, some member don't get calls from queues. In CLI: "queue show" shows something like: 611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no calls yet We use the extension 611 in different computers, in the

Hi folks -- I have a FastAGI Perl script running, handling calls. It works great. At one point I have a Dial() command. If the called party hangs up, Dial() returns 0, and when I call my own recordCdr() function using the channel variables ANSWEREDTIME, DIALEDTIME and DIALSTATUS, everything is fine. However, if the called party picks up, and then the dialing party hangs up Dial() returns -1,

Hi All, Asterisk 1.4.12 CentOS 5 My ISP account includes nearly 500 minutes of VOIP calls per month but the service is expensive for unbundled minutes. So I'm trying to find a way to keep an accumulated total of calls made through that trunk so that I can automatically switch to a lower-cost provider when my bundled minutes are used. The plan is to store the accumulated time in AstDB and

Hi All, Asterisk 1.4.12 on CentOS 5 I'm trying to increment an AstDB key with the length of the last outgoing call. Here's what I've got for "01" UK geographical numbers: exten => _01.,1,Dial(${UKGeographical}/${EXTEN},,g) exten => _01.,n,Log(NOTICE,Call to ${EXTEN} lasted ${DIALEDTIME}) exten => _01.,n,Set(CALLTIME=${DIALEDTIME}) exten =>

Hi, I noticed that when dial terminates it does not return to the dialplan, and therefore can not execute any entry after Dial(). Is there any trick to overcome this limitation ? How I am supposed to handle the returned vales DIALEDTIME, ANSWEREDTIME if I can not execute anything after Dial()? I made a workaround with DeadAGI (below) but it is unreliable: if 2 calls end

I'm looking into the dialplan specifics: tleilax:~ # tleilax:~ # cat /etc/asterisk/extensions.conf [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo TRUNK=DAHDI/r1 ; Trunk interface TRUNKX=DAHDI/r2 ; 2nd trunk interface TRUNKIAX=IAX2/ASTtest1:test at 10.10.10.16:4569 ; IAX trunk

Hi, I have a dial plan where I need to notify an external system when a call was answered and when the call hung up. In both requests the start time needs to be the same. My Dialplan looks something like this: [outbound] Exten => _X.,1,Dial(SIP/${EXTEN}@1.1.1.1,,U(call-answer-from-carrier)) Exten => h,1,NoOp(ANSWERED_TIME: ${ANSWEREDTIME} >>> DIAL_TIME: ${DIALEDTIME}

Hello, I was wondering if there is a simple way to limit the number of simultaneous calls in an Asterisk PBX ? I've seen that we can make this easily per channel (like in SIP.CONF) : incominglimit=X, but I'm looking to limit the maximum calls all channels together. Another thing. Working with asterisk-perl, I need to get the call duration, currently I use

Hello All. I am trying to use RAGI the ruby agi framework with trixbox. I am having a problem with the dialout part. The RAGI framework creates a file in the /var/spool/asterisk/outgoing directory and routes the call to an extension (I have listed the relevent portion of the file below). The problem is that the initial dial command does not execute properly in trixbox. I am hoping somebody who

READ READ READ .... http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics.html Regards, Mitul Limbani, Business Head, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mitul at enterux.in DID: +91-22-71967196 Cell: +91-9820332422 On Sun, Feb 22,

Hello, I have a strange h323 issue. After executing command "Dial("SIP/333-0d1dfe00", "H323/361737052390920 at ccg|5|tT")" at Oct 18 22:32:23. Meanwile I have sniffing traffic on port 1720. The call was established just at Oct 18 22:33:03 (New H.323 Connection created.) and also packet sniffer grabs the h323 invites at this time also. So my question is what

Hello, I installed Asterisk from CVS on Redhat Linux 9 and working with chan_h323 module and g729/g723 free codecs too. My network connection diagram: ---------------------------------------------- X-lite/X-Pro-->Asterisk--chan_h323-->GnuGK--->AS5300-->PSTN boldsoft*CLI> show version Asterisk CVS-v1-0-03/24/06-15:27:08 built by root@boldsoft on a i686 running Linux I can make

I'm haveing trouble connecting an Ericsson Webswitch 100 to asterisk. Has anyone gotten a Webswitch running? When I try to connect asterisk thinks everything works fine, while the webswitch just rings. I belive chan_h323 is picking the wrong port to talk at the webswitch on, however I'm not sure, nor am I sure how to fix it. Any clues/hints? A tcpdump is attached to show the session.

It seems that what ever I set in my answer handler does not show up in the hangup handler. In order to do billing I can't rely on the g option where the caller hangs up the call. Looks like I can either use h or a hangup handler along with the shared function. On Tue, Dec 26, 2017 at 4:40 PM, Eric Wieling <ewieling at nyigc.com> wrote: > Don't use an 'h' extension, use

Hi team, I am trying to find solution to hangup b-party call after 1 min with out disconnecting the call of a-party but following dial plan which is disconnect both the calls. Please suggest me the solution. [TB] exten => _X.,1,Wait(${INCOMING_WAIT}) exten =>_X.,2,Verbose(TB) exten =>_X.,3,Answer() exten => _X.,4,Set(mainLoop=0) exten =>

Hi All; Really I need to know why when using the "h" in the exten =>, then we use DeaAGI with it? I am using vicidial and I see this line alot, so I need to know how it work (when it will be executed): exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}) The question is: When

Hi, I'm developping some AGI in perl (5.8.6) on i386 using Asterisk 1.0.5. I want to get some variables such as DIALSTATUS and ANSWEREDTIME after a $AGI->exec("Dial", "dial_string"); but here is what i get actually: DIALSTATUS= DIALEDTIME=ANSWER ANSWEREDTIME=18 I searched the archives and saw that $AGI->verbose could mess the access to variables, but I don't use

Dear friends, I have compiled and installed h.323 in my asterisk. And I have a service from a H.323 VoIP provider who give me user, password and gatekeeper IP address. All configured. But when I start my asterisk i receive the following error and h.323 calls can not be making and/or receiving. [chan_h323.so]=> (The NuFone Network's Open H.323 Channel Driver) == Parsing

Hello. I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage to make outbound calls, but the communication drops off after 30 seconds or so. I'd really appreciate having some assistance from the mailing list on this issue. So, I'm having an Asterisk server behind a firewall and Zoiper softphones on SIP connecting to Asterisk on the same local area network. The

Actually I am facing a problem with H.323 (the standard and the ooh323) with Asterisk vesion 1.4.25 and I discover the following: 1) Using the standard h323 that come with Asterisk: The chan_h323.so it is not existed in the /usr/lib/asterisk/modules after doing the compilation and installation for (pwlib, openh323, /chanels/h323, asterisk), although make menuselect was done and the h323 channel