similar to: app_changrab, replacement for meetme and conference: returning to dialplan

Hi, We are running asterisk 1.2.16 and need to connect two channels which are already established. We are currently using app_meetme to achieve that, but we are sometimes unhappy, as app_meetme provides functionality that produces load that we do not need in our two party conferences. I figured out that there is an alternative called app_changrab.

I released app_changrab.c lastnight really late... It includes a way to hijack a channel and originate calls from the CLI. /b --- Keep Your Friends Close, But Your Enemies Even Closer...

Hi, I have a problem with polarity reverse on answer I use asterisk 1.4.30 linux kernel version 2.6.27 dahdi version 2.2.1 and analog card is Sangoma a400 with fxo ports this is my config [trunkgroups]

Client - secureCRT 3.3 outside the firewall (Checkpoint) Server - openssh v3.7 on an aix51 rs6k inside the fw The firewall lets in the first packet but blocks the second with the message: ssh 1.x not allowed. The connection gets reset. Here is the trace from the client: [SSH LOCAL ONLY] : Connect: 12.x.x.x:22 [direct] [SSH LOCAL ONLY] : StateChange:

Hi, I just got my local yum repo setup so that I can use puppet to install packages... and it looks like something is not working. <snip> Tue Sep 19 02:13:05 PDT 2006 Puppet (notice): Starting configuration run Tue Sep 19 02:13:10 PDT 2006 //default/java/package=jdk/ensure (err): change from absent to latest failed: Could not update: Could not execute ''/usr/bin/yum -y install

I've tried a number of solutions, but I've been unable to get Asterisk working with streaming MOH without running into the "buffer" issue. I've tried using various combinations madplay, mpg123, mpg321. I've also tried streamplayer by itself, and in combination with play-fifo ( http://www.freeswitch.org/asterisk_stuff/play-fifo.c ) to try and eliminate the issue. For

Hi All, I have been running a environment with asterisk 1.4.20.1 for some time now with no issue but have recently added some extra functionality (enabled call recording via MixMonitor) and ran into some deadlock issues which seem to be well documented with earlier 1.4.x releases so have decided to take the plunge and upgrade. I decided to start testing with 1.6.2 but have run into a couple

Hi All, I am using an Asterisk 1.6.2.6, and when I use this part of the dialplan: exten => 8355,1,Dial(SIP/${EXTEN}&IAX2/${EXTEN},18,tTWwr) exten => 8355,n,Dial(IAX2/8366,48,tTWwr) (i made that simple to exhibit issue) I got just 1 ring in 8366 extension before it hangup, what i noticed is the total time spent on ringing is 30s that means if i use 12s in the first dial i get 18s left

Dear All, I just setup the Asterisk with E100P which it's no problem in Dial In but I have problem when outdial. The connection method is like this : E1 PRI <-SIGNAL-1-> MaxLink (PBX) <-SIGNAL-2-> E100P <-> Asterisk <--> SIP \-----> Analog PHone Now when I tried to dial out by SIP X-Lite on Windows, it shows me Connect, Trying,

I've got the ztdummy kernel module loaded and seem to have all the desired prerequisites in place, but Asterisk never seems to compile with MeetMe() application support enabled, nor does there appear to be a module I am failing to load that would contain this application. Is there something really obvious I am missing? Thanks, -- Alex Balashov Evariste Systems Web :

Hi, We are trying to use attended transfer with Asterisk 1.2.5, but when we do the transfer and dial the new number, it times out after 3 rings and then the callee is put back to the original agent. Where can I adjust the timeout which applies to the number we are transferring to? I have changed the extension for this number to timeout at 60 seconds, but that seems to make no difference. --

Hi, I am having some trouble with dovecot (0.99.14) and maildir. Some users are reporting that sometimes, they don't get (pop3, various clients) new mails for a while even though they should. I checked and confirmed this: mails were delivered by MTA (sendmail, procmail). But when the problem existed, the new mails were in cur/, not new/ and dovecot would not report any new messaged to the

Hi all. I have enabled the attended transfer feature in features.conf. I'm using it and I want to resolve some questions, I hope someone can help me :) When I transfer a call to an extension: - The extension rings during 15 seconds and the call returns to the "transferer". Is there any possibility to recover the call before the timeout of 15 seconds expires? I mean, I would like

100% repeatable (for me). Sip phone A calls Sip phone B. Either Sip phone A or B does #700. The party that keyed #700 gets the parked announcement (eg 701) and the other party get MOH. There is still an audio channel between the two SIP phones at this point. When the party that typed #700 hangs up, Asterisk crashes. This has been working in previous 1.4's (but not 1.4.10) and I

In the past, there have been reports of problems with Asterisk with multiple processors and/or HyperThreading. I'm having a !@#$ of a problem with an HPDL380 with 2 3.4gHz Xeon processors, 2 gb RAM -- if I got 24 hours I'd think I had died and gone to heaven :) Am I missing something obvious like "Asterisk is single CPU, single core?" I can't access the ILO so I

Hi all, i'm testing my asterisk and without warning i can not hear any audio file (the files situated under /var/lib/asterisk/sounds). I don't hear no audio and i get this message on CLI: *CLI> -- Executing Dial("SIP/2339-4e1d", "SIP/2391|60|Ttr") in new stack -- Called 2391 -- SIP/2391-d264 is ringing -- SIP/2391-d264 answered SIP/2339-4e1d --

Group I'm having some trouble with asterisk and the page cmd. Any help would be great! This is what's in my extensions.conf exten => _**2,1,SIPAddHeader(Call-Info: answer-after=0) exten => _**2,2,Page(SIP/36651)|d exten => _**2,3,Hangup CLI output ******************** Connected to Asterisk SVN-branch-1.4-r57207 currently running on VoIP-PBX (pid =

I tried to do what I with regular SIP to Transfer a call via 302 Redirect. In asterisk 12 we need to add the Tech, or not, but in any case, there is no transfer done. The call is closed. Here is a trace. How do I do this? [Jul 9 21:39:29] DEBUG[47716][C-00000002]: pbx.c:4869 pbx_extension_helper: Launching 'Transfer' -- Executing [17274428141 at redirect:30]

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

Hi, i use chan_capi 0.3.1 with asterisk (stable branch cvs) and 3 x c4 active ISDN card. From Controller 1 - 7 there are no problems making calls between asterisk and the pstn. But when i make calls from controller 8 - 12 i get on every controller (8 - 12) a segmentation fault in asterisk :( I tried different linux distributions (gentoo 2004.1, redhat 9.0 , suse 9.1) but same error.