similar to: Problem in extensions.conf Configuration ${CALLINGPRES}

I've just upgraded to the latest CVS head, and my outbound calls stopped working. I traced it back to the line exten => s,9,CallingPres(${ARG2}) It seems as if this application is now missing. I tracked back the changes and found in 1.415 of chan_zap.c the code was removed because it was "duplicated". However, it does not exist anywhere ! Am I being stupid, missed

Dear Max, Yes, I have modified line no 5 with include_once(dirname(__FILE__) . "/lib/fpdf.php'); But I can not also see export to csv or export to pdf option from download cdr data as I required within date. -- With Regards, Hiren Mistry

From the Wiki: Presentation indicator (octet 3a) Bits 7 6 Meaning 0 0 Presentation allowed 0 1 Presentation restricted 1 0 Number not available due to interworking 1 1 Reserved Screening indicator (octet 3a) Bits 2 1 Meaning 0 0 User-provided, not screened 0 1 User-provided, verified and passed 1 0 User-provided, verified and failed 1 1 Network provided How do these bits fit into

greetings, would it be a good idea to include within the scp man pages, how one would use scp to copy files which have the ' : ' character in the filename? a friend struggled a little trying to do this, discovered the answer on google, i thought it might be usefull to have in the man pages too. the solution was to use the files' full path so that scp recognised it as a file. please cc

Hi, I'm using the macro below in extensions.conf for most of my outbound calls. One issue with my current configuration is that when I make an outbound call it doesn't properly detect that my PSTN line (Zap/1) is busy with another call and then overflow to my outbound IAX connections. I think the root cause is that DIALSTATUS gets reported as BUSY. The debug output is below. My desired

I can dial from 601 to a public number. The public number rings. I pickup and hear nothing, while on 601 it keeps ringing. (BTW, is it right to say "ringing" on the active phone?) The *CLI> doesn't show me anything useful: Executing Macro("SIP/601-8238", "dial-pstn|88097880|") in new stack Executing SetGlobalVar("SIP/601-8238",

Hello, this is an example extensions.conf. [default] exten => 500,1,Answer exten => 8,1,SetGlobalVar(firstdigit=8) exten => 8,2,Goto(process,s,1) exten => 9,1,SetGlobalVar(firstdigit=9) exten => 9,2,Goto(process,s,1) I call extension 500 and send dtmf digit 9. This is printed to the CLI: -- Executing Answer("Zap/20-1", "") in new stack -- Accepting

On Thu, Jun 6, 2013 at 11:58 AM, upen <upendra.gandhi at gmail.com> wrote: >> This is sounding like dropped/intercepted packets. Could we get some >> details on the client (make/model of computer/motherboard, same for >> NIC, PXE OROM details)? Is it connected to the same physical switch >> as the DHCP/TFTP server? Any chance that the physical switch that the

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

Helo List, First I would like to apologize for my bad spelling as well as that I did not search the wiki first. I only have email access at the moment. I am having trouble setting both variables and global variables thru an extension. I am using Asterisk 1.2.4 with Ztdummy on CentOS 3.4 with an Xlite softphone. I have two xlite phones on diffent computers. One logs in as xlite1 and the other as

I got two phones on an ATA-186 (601, 602) and two phones on the TDM22B (603, 604). I have two lines on the TDM22B. I cannot figure out some of the problems: 1. 601 dials via ZAP/3-1 to local phone number at PSTN: ringing pickup on PSTN (empty) still ringing in the phone set 601 2. call from PSTN back: 601 picks up ... everything works !!! No caller id shows up 3. For testing I have only one

Hi guys Ive been trying to get this to work for ages now, basicaly im trying to do if ${woteva} = "" (nothing), or its none existenant then do label 1, else label 2. for my last called function, so it will play a different message if theres no last call in the system or it was anonymous. ive tried exten => 1000,1,GotoIf($[${last-call${CALLERIDNUM}} = ""]?4:5) and heaps of

Hi there, I want to set up an extension set that acts different depending on the count of used lines. I have a EuroISDN E1 board with mISDN and I only want to offer 10 lines. Therefore I set up a global variables LINES in the general section of extensions.conf and instantiate it with 0. I a call is incoming I check the LINES variable wether is 10 or more. If so I make a call transfer. If not

Hi all, I need to test the following scenario: +-----------+ +-----------+ | asterisk 1| | asterisk 2| +-----------+ +-----------+ | | | | _______|__________________|___________ | | | | | | +-------+ +-------+ | ATA 1 |

Hi I am using Asterisk 1.2 and for the life of me, I am unable to transfer outbound calls (eg calls I initiate from sip extensions). When I press #, nothing happens. Inbound calls transfer fine, but only once per call. The problem happens: - With both software and hardware phones. - With calls going out through the ZAP channel and to internal SIP extensions. - After I have transferred an

Hi, I am trying to pass DIALSTATUS to a Macro so that i can set a variable when a call is placed (call is placed via a call file to another extension first). Basically i don't want to dial a number where a call is already bridged and thats why i am setting a variable. [macro-afterdial]; exten => s,1,Goto(s-${ARG1},1) exten => s-ANSWER,1,SetGlobalVar(NUM${ARG2} = "ACTIVE")

Hi. Can anyone help me with the following: [globals] OFFICEHOURS .................................... [internal] exten => *80,2,SetGlobalVar(OFFICEHOURS=100) exten => *80,2,SetGlobalVar(OFFICEHOURS=200) .................................... [incoming] exten => s,1,GotoIf($[${OFFICEHOURS} = 100}]?incoming-officehours:incoming-officehours-off 1. Am I using the right sytanx when

On 09/05/16 at 12:57P, Slawa Olhovchenkov wrote: > I am try using 11.0 on Dual E5-2620 (no X2APIC). > Under high network load and may be addtional conditional system go to > unresponsible state -- no reaction to network and console (USB IPMI > emulation). INVARIANTS give to high overhad. Is this exist some way to > debug this? Can you panic it from console to get to db> to get

Hello, I have an icecast2 server streaming some mp3 files and am using winamp to listen to the stream, however I couldn't figure out if there is a way to seek or rewind. Is there a way to seek/fast forward/rewind/pause using icecast2 ? <p>Regards, -- Hiren --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe

On Fri, 2018-07-20 at 10:10 -0500, Michael Hennebry wrote: > On Tue, 17 Jul 2018, Pete Biggs wrote: > > > Check your hardware - all these things might well be symptomatic of an > > hardware issue. > > I'm not at all sure how. > If it means opening the case, > hardware issues are likely to occur. > > Is there a way to tell whether the video > player is