similar to: [1.4.21.2] Read() disconnects half-way through?

Hello, I need to write a script which prompts the callee to type a number, and then read it back to them as confirmation: ======= extensions.conf [robocall] ;Expect 10-digit number excluding final #, 2 tries, 20s time-out exten => s,n(nbr2call),Read(NBR2CALL,please-type-number,10,,2,20) exten => s,n,GotoIf($[${LEN(${NBR2CALL})} != 10]?end) ;exten => s,n,SayDigits(${NBR2CALL}) exten

Hello I'm trying to use ChanIsAvail() to check when the landline is back to idle after a call, but for some reason, Asterisk doesn't detect that the callee has hung up after listening to MoH for a few seconds: ========== extensions.conf ;Play MoH for a few seconds, hang up, and ;check ChanIsAvail() able to detect when line idle again exten => 8888,1,Answer() exten =>

Hi I just got an AA50 from Digium and the paging command reboots asterisk when you use it. Digium says it is a requested feature and is of low priority. Is there any other way to page 10 Grandstream gxp2000 phones with meetme or some other command than the page command. Thanks in advance. Kelly -------------- next part -------------- An HTML attachment was scrubbed... URL:

All the aforementioned techniques need change everytime on the dialplan. I need the office secretary to edit a file (call file) and place it in a particular folder in their windows PCs. this folder is the outgoing folder of LINUX shared through samba in LAN. i need to make it as easy as possible, please. On Tue, Mar 20, 2018 at 5:41 PM, Frank Vanoni <mailinglist at linuxista.com> wrote:

I'm using Asterisk 1.2.7.1. Has the callerid number matching functionality in extensions.conf changed recently? exten => 5555,1,NoOp(${CALLERID}) hestia*CLI> -- Executing NoOp("SIP/2944093-d24d", ""Cletus the Slaw Jawed Yokel" <2944093>") in new stack == Auto fallthrough, channel 'SIP/2944093-d24d' status is 'UNKNOWN' This

Hi All, I am new to Asterisk. Now I got one question on the identifier parameter of the Dial() command. I saw as below: exten => 20,1,Dia(Zap/3/5551234). Would you please let me know the meaning of "5551234"? Thanks, Songtao -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am trying to implement the callback feature of our old phone system. This feature may go by a different name in asterisk? It worked as follows. If phone A called phone B and it was BUSY, you press a button to enable a callback. User A is free to continue work or make other calls. What this meant is that when both phones became free, phone A would ring, on answer it would call phone B

Hello For some reason, when dialing out through a call file and the remote line is busy, Asterisk doesn't jump to the "failed" extension in the context used by the call file: ====== call file Channel: Zap/1/5551234 Context: callbacktest Extension: start Priority: 1 MaxRetries: 1 ====== extension.conf [callbacktest] exten => start,1,NoOp(Status is ${DIALSTATUS}) exten =>

Hi, I've created a test at "extensions.conf" like this with 3 steps: ; When dial 5555, get the first available channel and dial do 482343400 exten => 5555,1,Dial(Zap/g1/482343400,5,rt) ; When dial 5555, get the channel 20 and dial do 482343400 exten => 5555,2,Dial(Zap/20/482343400) ; Go to Voicemail 1234 exten => 5555,3,Voicemail(u1234) I've tried using just the

Just throwing this out here, hopefully someone can tell me why. *CLI> show version Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running Linux Zap/g1 is pri_cpe to Bell Canada 5551234 is a normal POTS line I have busied out (handset offhook) exten => 1234,1,Dial(Zap/g1/5551234,,g) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is

Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890

Hi, i tried to compile Asterisk 1.4.21.2 on a server which i have been using with many previous Asterisk versions, without any problems. But with 1.4.21.2 it failed: ---------------------------------- [CC] res_adsi.c -> res_adsi.o [LD] res_adsi.o -> res_adsi.so [CC] res_agi.c -> res_agi.o [LD] res_agi.o -> res_agi.so [CC] res_clioriginate.c -> res_clioriginate.o

Hi. in my system i have a conference room where someone can call it eg 698 dial the PIN eg 1234 and enter the room as a user. The admin enters in through a different number and PIN. I would like to have a call file and call all participants eg 610-619 at certain time of the day and give them access to the conference. During my try i managed to create a call file where it calls the a SIP phone and

Hi list, I just experienced an odd behaviour in 1.4.22 vs 1.4.21.2. To cut a long story short, IAX2 is not tx-ing hangup... Scenario is composed of two asterisk systems A and B. A receives calls from IAX users X, Y, Z, etc, does some validation and forwards them to B, also over IAX. When B hangs up, it transmits IAX hangup which A receives who, in turn, does not transmit the IAX hangup to its

This wav produces a bit of audible static when encoded at the highest bitrate vorbis will allow me to encode at (30.7kbps avg.): http://staff.xmms.org/zinx/misc/5551234.wav.gz An encoded version, with the bitrate set to approximate 128kbps (I think it output to 29 some odd kbps): http://staff.xmms.org/zinx/misc/5551234.ogg This is with the latest CVS tree as of Sat May 12 10:06:17 UTC

Hi, After upgrading our server from asterisk 1.2.16 to 1.4.21.2 we experience crashes at random intervals with: [Nov 6 11:03:28] WARNING[12230] app_dial.c: Unable to forward voice frame read(0, <unfinished ...> +++ killed by SIGSEGV (core dumped) +++ Process 15755 detached On a second sister-machine with a mirror install we have the same problem. So it doesn't seem to be a

I am investigating an issue with IAX2 extensions in Asterisk 1.8.x. My application connects to Asterisk via AMI and attempts to run an Originate command between an extension (such as SIP/5555 or IAX2/8888) and an application (in my case it is AgentLogin). This works correctly for SIP extensions, in all Asterisk versions. With IAX2 extensions, this worked correctly in Asterisk 1.6.2.20, but

Hi, Ii try to connect an Asterisk server running 1.4.21.2 version with gtalk2voip services. Everything is fine till the call for DTMF test: there is no audio and Asterisk shows [Nov 18 14:51:47] WARNING[20502]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission SIPCALL-435578583-1984100284 at 72.20.112.114 for seqno 1 (Critical Response) [Nov 18 14:51:47] WARNING[20502]:

Hi guys.. I just wanted to know if this config could work correctly, since a lot of u guys have been working a lot with asterisk... Asterisk 1.4.21.2 -> chosen because is the last with Zaptel support ? Zaptel 1.4.12 Add ons 1.4.9 Currently i have a 1.6.1.0 Server running with no problems, but I'm about to buy a openvox a400p Card, so i would like to test both environments, 1.4 and 1.6,

Hi all, I want to test the Native Bridge mode of DAHDI (FXS/FXO). I use asterisk 11.14.2 and DAHDI 2.8.0. I try to set callwaiting = no AND callwaitingcallerid = no in chan_dahdi.conf. But I can't find native bridging information from CLI(opened debug mode in logger.conf). How can I test the dahdi_bridge in native bridge mode? I use normal dial command ex: Dial(DAHDI/2,30,tTkK) to dial from