similar to: Sharing variables between contexts

Hello all, First of all, let me apologize about the length of this message, but I suppose it was necessary to include the details. I've spent quite some time already trying to get the call transfer function to work on my Asterisk installation. Let me first describe the general situation of the setup I am using, so you might be able to pinpoint the cause of the problem. I'm currently

hello like if 6000 is the main exchange number. any one dial to 6000 will be asked for pressing his desired extension then he can press his desired extension then his number is diled exten=>6000,1,Background(enterdesiredexten) exten=>6000,2,Wait(2) exten=>2000,1,Dial(SIP/${EXTEN})

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Sorry, i also tried this: exten => 6000,1,Answer exten => 6000,2,MusicOnHold(default) and i got this result: *CLI> -- Executing Answer("SIP/2391-8cdd", "") in new stack -- Executing MusicOnHold("SIP/2391-8cdd", "default") in new stack Jun 29 19:33:47 WARNING[1616]: res_musiconhold.c:354 moh0_exec: Unable to start music on hold (class

I am calling from my laptop to an asterisk box which answers the call and I can hear the voicemail prompts, but the problem is that after so many seconds, MSN Messenger drops the call because it thinks it hasn't been answered by the remote machine. I'm not sure if this is an asterisk problem, or if it is Messenger not knowing the call was answered. Has anyone else run into this sort of

The wiki says this about the MusicOnHold command: "Plays hold music specified by class. If omitted, the default music source for the channel will be used." http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold How do I set the default music on hold class for the SIP channel ? I tried adding musiconhold=test to my sip.conf. musiconhold.conf looks like this:

Hi, Updating zaptel gives me this during the make. Any ideas, the searches and Wiki gives me no hints. In file included from /usr/src/linux-2.4/include/linux/fs.h:19, from /usr/src/linux-2.4/include/linux/capability.h:17, from /usr/src/linux-2.4/include/linux/binfmts.h:5, from /usr/src/linux-2.4/include/linux/sched.h:9, from

This is my musiconhold.conf and my folder: root@voip:/etc/asterisk# less musiconhold.conf [classes] default => quietmp3:/var/lib/asterisk/mohmp3 ;loud => mp3:/var/lib/asterisk/mohmp3 ;random => mp3:/var/lib/asterisk/mohmp3,-z ;unbuffered => mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3 ; Note that the custom mode cannot handle escaped parameters

Any thoughts on the following? I am running asterisk from CVS (downloaded yesterday's version, just to be sure) on a test system with no digium cards in it, so I have installed ztdummy (see logs and screenshots below) as a timing source. When I call the music on hold extension from a Sipura Sip connected analog phone, I hear nothing and start getting Warning[98310]: chan_sip.c:674

I'm testing a few IAX trunk scenarios in a controlled lab. From server2 extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes across the IAX trunk to server 1 (IP address 172.16.200.210). Instead of ringing the 6001 phone, it plays tt-weasels (the s extension). When I dial 6099 it also plays tt-weasels as it's supposed to, but it's not the tt-weasels under its

There is an option "--reverse" to run examples in the reverse order which is good for detecting inter-dependencies between examples. Is it feasible for rspec to automatically switch between "forward" and "reverse" order each time a spec is run? My goal is to detect inter-dependencies sooner than later... Thanks Alvin.

On 17.06.2016 16:27, ????????? ???????? wrote: > Walter H. ????? 2016-06-16 22:54: >> On 16.06.2016 21:42, ????????? ???????? wrote: >>> >>> I don't think OCSP is critical for free certificates suitable for >>> small businesses and personal sites. >>> >> this is philosophy; >> >> I'd say when you do it then do it good, else

All, On our main file server, all our home directorys are automount points. So /home isn't a real filesystem area, it points to a bunch of different home areas based on the auto.direct table. Example: /home/user1 mounts from --> /export/home1/user1 /home/user2 mounts from --> /export/home2/user2 /home/user3 mounts from --> /export/home2/user3 /home/user4 mounts from -->

I can't get 3.0.1 to fully compile on my Solaris machines(2.6 and 2.8) Below is an example the error I am getting. Note, that it errored out earlier with the same problem, I typed make again and it went on to the next file, got the same error about a file with extension .o.o not existing. The actual error: --- Compiling lib/replace1.c with -KPIC mv: cannot access lib/replace1.po.o.o ***

Guys I am having some serious issues with my call queue and Management is breathing down my neck pretty bad, and I am running out of ideas. I have a single queue for my tech support department. I originally was using the AgentCallbackLogin for them and it tested out great on our testing weekends, but it hasn't worked out since. It would only let one of them take calls at a time, no matter

Not sure if this is the correct forum for this, but I've found what I would consider to be a potentially serious bug to the unsuspecting user. Given a numeric vector V with class labels in R, the following calls 1. > sort(as.matrix(V)) and 2. >as.matrix(sort(V)) produce different ouput. The vector is sorted properly in both cases, but only 2. produces the correct labeling of

Hi All... I have a need to phone a large number of people and collect information from them. I know Asterisk has a nice IVR system, but can it be used to initiate a call to people listed in a database or text file? Don't worry, this is not an annoying marketing thing. Thanks...

Hi everyone, We are working on enhancing our internal testing so we can improve the overall quality of Puppet releases. For example, I am one of two QA engineers assigned full-time to testing Puppet''s FOSS stack. I''m seeking your input and experience to help us identify what we''re missing and what we can improve. We have an extensive test suite, but there''s

Hello All, Does anyone know if you can start an MOH queue on an individual call? What I mean is, for example if you have a script that you want the moh to start with certain phrases, can it be done, or are you limited to the standard looping audio? It's almost like starting a stream for each call, and terminating it when the call comes off of hold. Regards, Greg

I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines connected via TE405P. Everything works great, except MOH. I added an exten with MusicOnHold(30), and it plays just fine. Conferences have music when no one is in. I have SIP phones. When I place a call on hold, the CLI give no indication the call is on hold. I have set musiconhold(default) everywhere, removed it from everywhere,