similar to: RE: Asterisk-Users digest, Vol 1 #3368 - 12 msgs

Is it true that every time we make a change in the configuration file we need to restart the asterisk server. This will not be practical in the production environment. Thanks,

Hi, I tried to make my Linux box smbmount a remote share originating from a Macintosh running the commercial package DAVE 2.5. The Linux box is a RedHat 6.0 with a 2.2.14 kernel (with smbfs inserted as a module) and samba-2.0.3; the iMac ran MacOS9. The directory to be shared, \\akemi\archivio, contained two files, aaa and bbb. Here's what happened: [root@sonal ~]# smbmount

Hello, This is what what I see after a Yum install asterisk16 asterisk16-config freepbx: Use of uninitialized value in string ne at /var/www/html/panel/op_server.plline 4997. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5439. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5440. Use of uninitialized value

Hi there, We did a clean install the AsteriskNOW 1.7.0 64 bits ISO and configured it. On the main screen (Crtl-ALT-F1) we keep seeing the following lines when making a call Use of uninitialized value in hash element at /var/www/html/panel/ op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372. Use of uninitialized value in

I am a newbie here and I have been going around in circles with this problem. I have set up samba to use encrypted p/w and use a WinNT4 sp4 client with registry changed for plaintextpasswords. I can 'map network drive' successfully and peruse all the shares but if I try to use the 'Network Neighborhood', then I can see the shares (so I guess that I have been authenticated

Hi there, I'm wondering if there is a way to assign a different Caller ID to each Zap interface. I have 3 Digium X100P cards, and I'm sure there must be some way of configuring zapata.conf to allow each line to identify itself with a different Caller ID string. Many thanks, Steve -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077 7338

Hi there, I'm having some trouble with getting Asterisk to make a call, I think it should be quite easy, but anyway... Using the following file contents: ## Channel: Zap/3/<TEL NUMBER HERE> MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: phones Extension: 502 Priority: 1 ## Extension 502 is simply one that plays a sound back. When I dump this file into

When I drag and drop a call from the PSTN to a SIP phone (the SIP phone's icon) using FOP .25 with * 1.0.9 with the intent of transferring it, the called party gets transferred rather than the calling party. This is controlled by the reverse_transfer parameter in op_server.cfg but the behavior is exactly the same whether the parameter is set to 0 or 1. This is after the call is picked up by

Hi there, Just wondering if there is a way to speak to the person you are transferring a call to before actually connecting them to the incoming call. E.g. "Hi, Colleague, I've got Bill from Microsoft on the line here... putting you through now" Then actually transfer the call. Does that make sense!? -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077

This strikes me as something that should be really very simple to do, but I can't figure it out. Is there a way of logging all SIP debuging info to a file somewhere? It would help me greatly! Cheers, Steve -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077 7338

Hello All I am looking for a PC based switchboard application. Cisco CallManager has a web attendant console that allows you to use the PC to transfer calls and the like and I was wondering if there was a similar program compatible with *. Thank you in advance Keith D'Atrio -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've got a fairly working Asterisk setup, with a few minor glitches, one of which is very very irritating. Sometimes, during a call, the remote end just drops off. We're using software SIP phones (SJPhone) connecting to * then out through analogue lines with X100P cards. There is nothing in the logs and nothing on the console, the call just seems to 'go away'! Can anyone

Hi, How can we plot two graphs ex. lets say correlation & ratio in the same window? I mean in the window I have : 1. Graph of correlation having X & Y axes 2. Graph of ratio having A & B axes one above the other. Thanks, Sonal

Hello All, I'm new to the list and the whole voip server side. I'm trying to setup Asterisk to just do internal dialing, no access out to the pstn is required/wanted at the moment. I'm running Fedora Core 3 with Cisco 7960's phones (running SIP 6.3). I've set it up following these guides: http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3

hi, i am a new member. i am using R in finding correlation between two variables of unequal length. when i use cor(x,y,na.rm=T,use="complete") where x has observations from 1928 to 2006 & y has observations from 1950 to 2006. I used na.rm=T to use the "complete observations". So missing values should be handled by casewise deletion. But it gives me error Error in

Hi - I've told lots of people about the Flash Operator Panel before, but I've never actually used it myself. I've got it all set up nicely, but I can't seem to authenticate to the asterisk manager (which is running on the same box). When I set the op_server.pl to give debug messages, it shows that it can reach the asterisk manager, but cannot authenticate: ** Asterisk

Hi there... I have a simple * setup with about 11 Soft phones (SJ Phone). The clients don't support a consultive or supervised transfer (I believe that's what it is called). Tris is a feature much desired by the powers that be and they want me to "make it work" :) I was wondering if there was a way to do this with and AGI script or the like so that when Staff 1 gets an external

Hi, How do we find out the inner product & norm of eigen vectors in R? Lets say we have eigen vectors : x1 = 1,2,3 and x2 = 2,-3,4 are there any functions buit in R which directly calculate the inner product & norm of vectors? Thanks, Sonal. [[alternative HTML version deleted]]

Hi, Anyone got AFOP working. The install instructions tell you to copy all of the files extracted under the 'html' directory to a subdirectory under your main web directory (in my case this is /var/www/html/panel/) and then the instructions talk about modifying the 'op_server.cfg' file but they do not tell you were to put this file. There is something wrong with the

Hi Guys, Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When trying to dial a number, I get this: tel*CLI> Use of uninitialized value in hash element at /var/www/html/panel/ op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372. Use of uninitialized value in pattern match (m//) at