similar to: Read command

Hello, I'm using X-web lite in a webpage to connect to one of our asterisk server. But now I have a problem, when you are connected to a voice script the voice will not be heard after a couple of seconds. When you press or say something that the voice will come back for a couple of seconds. When I thy X-Lite (stand-alone version) I had the same problem, but when I turned off the

Hello, To connect to our oracle database from an asterisk application we use res_perl. Sometimes one of our asterisk server will 'freeze' and work anymore. I have to kill the job safe_asterisk and start it again, so that the application asterisk works again. If I look in the log files it look like that asterisk will 'hang or freeze', if two callers calls exactly at the

Hi, Does anybody know if RAMI (Ruby Ami) is still functional? And is this still compatible with asterisk 1.8 Best Regards, Arjan Kroon Mobillion BV

Thankx MC, This is the solution. I've tried it and it works perfect. But I've got a question. I want to set a variable with the command SetVar I place the following text file in the directory /var/spool/asterisk/outgoing/ Channel: Zap/g1/0655871460 MaxRetries: 0 RetryTime: 30 WaitTime: 30 Context: call_outbound Extension: s Priority: 1 SetVar: call_outbound_id=0

Hai, Does anybody have problems with a wrong Connected Line ID with asterisk version 1.6 The following bug was for version 1.4, but I cannot make up if this bug is still in version 1.6 http://forums.digium.com/viewtopic.php?t=7780 In version 1.8 it is possible to change the Connected Line ID, but this isn't the case in version 1.6 Regards, Arjan Kroon Mobillion BV

Depends what you want to do! Do you want to do VoIP over that T1 to a provider or IP telephones? Do you want to hook up to the PSTN through that T1 as 24 voice channels, through a T1 card on your asterisk? If you want to use the T1 as 24 voice channels, the Telco is going to have to re-provision the T1 as a voice T1, because currently, presumably it is one big channel of data. You could have

Hi, How do you all handle the situation when a centrale fastagi server process(es) are down? AGI(..) prints "Unable to locate host" and the dailplan jumps to extension h. I'd like to handle the return value and keeping the caller in the dailplan and not to the hangup extension. Any tips about how to handle a AGI(..) returns -1 condition? thx Arjan Kroon Mobillion BV

Hi, We encounter a problem with the variable CALLERID(dnid) We use E1 lines where we can make an inbound call or an outbound call on the same channel (not at the same time) If the CALLERID(dnid) is not used, than the CALLERID(dnid) will be the CALLERID(dnid) of the previous call For example: - First we get a inbound call on channel DAHDI/11-1 with CALLERID(dnid) = '655871460' We read

Hi, I'm using frequently the perl api within asterisk. Now I'm looking for documentation for the perl commands. Some perl commands I found on this URL: http://www.voip-info.org/wiki/view/Asterisk+PHP Does anybody got more documentation or where I can found some more documentation about perl commands Kind Regards. Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus

Hi, We are using asterisk version 1.0.0. For queue'ing we use the rawplayer script to play a music file in the background. Now we see that after a while all the sessions on our Linux environment will be taken by the rawplayer process. An example of such a session is (done with ps -ax|grep rawplayer) 24785 ? Z 0:00 [rawplayer <defunct>] 8415 ? Z

Hi, Is it possible to use a live audio stream in asterisk I want to call a number and then hear an external audio stream. For example http://www.radioveronica.nl/radioveronicaplayer/radioveronica.asx I thought it was possible to use musiconhold, but I did not get it working. This is my musiconhold.conf ; ; Music on Hold -- Sample Configuration ; [general] [default] mode=custum

hello List, i am in a situation where i cannot get cdr records for call originated from CLI , i am not able to get when i used application or extension. is there any solution regarding this ,i working since last 3 days onto this. regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Can anybody tell me which version of res_perl I have to install on Asterisk 1.2.4. I tried to compile res_perl version 3.5 on Asterisk 1.2.4 and I got the following error. gcc -Wall -DRES_PERL_BASE="\"/usr/local/res_perl\"" -DMULTIPLICITY - D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS -fno-strict-aliasing -pipe -Wdeclaration-after-statement

Hi, I'm using videocalling on asterisk 1.4.10. When I setup the videocall with exten = n,1,h324m_gw(s at video2webanswer), I loose the variable DNID (${CALLERID(dnid)}) Before the videocall is set up, this variable is filled and after this videocall this variable is empty. Also all local variables are empty. If al look at the A-number (${CALLERID(num)} this variable is not empty

Hi, In my application I jump to different extensions For example: [begin] exten => s,1,Goto(starts,s,1) [start] exten => s,1,Play(welkom) ..... exten => h,1,Goto(end,s,1) [end] exten => s,1,Macro(end_call) exten => s,n, Hangup When I look at my CDR record I see three different CDR's in my record. Is there a way to use one CDR on every call, and not

Hey there, I'm trying to convert some call recordings from asterisk we have in .gsm format to something I can pipe through ffmpeg - wav would be good, mp3 would be amazing! I've been trying playing with sox but I don't seem to be getting too far with 1239101491.30.gsm -ql -r 64000 -t wav 1239101491.30.conv.wav resample as ffmpeg borks at it: tim at freee-meee:~/dmc/call

Hello, I'm a newbie in connection several asterisk servers with each others. I've got the following situation. I've got 9 asterisk servers (asterisk00 till asterisk08). When I call to asterisk08 then I want to redirect an application which runs on asterisk00. But how can I redirect in an application on asterisk08 to an application on asterisk00? Or isn't this possible?

Hello, I want to record the message of both parties during a dial. My extensions.conf at the line where dial is looks like this: exten => s,803,Dial(SIP/arjankroon2,30,rR) My Sip.conf look like this: [arjankroon2] ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend

Hi all, Yes, this is not the right list for such a question and I am using Asterisk myself its for a friend who isn't used to Linux. You can write me off list if you want. He is looking for a Windows based PBX with same functionality as Asterisk. Any tips? Many thanks!

Hello, I'm trying to get the following situation. Someone calls an application on one of our asterisk server. In this application the caller will call a SIP client. (with the command Dial) The Sip client is connected with another asterisk server. (see below) Caller --> asterisk01 (incoming server) --> asterisk00 (outbound server) --> SIP client (X-lite) Do anybody now how