similar to: 'System' application exit with error even if it performs the job as expected

I thought I would post this in case it might be of any use to anyone. Not anything special but it does work. Keep in mind you need sox and wmix. Here is some relevant exerpts of my extensions.conf using John Todds macro. [globals] CALLFILENAME=foo FOO=foo CALLERIDNUM=foo [default] exten => 287,1,Macro(dial,SIP/agent20002|20) exten => 287,2,Voicemail(u287) exten =>

Hi, I have two Asterisks identically installed on two computers. One of them is directly connected to the Internet, the other one through a NAT router (Netgear MR314). On the one behind the router I have an X100P card installed for PSTN connections. In the local LAN of each PBX they are several hardware IP phones (Cisco 7960 and 7940 with SIP images, firmware image P0S3-04-4-00.bin). I have

I have a question on how to handle the "h" routines. I have noticed that if the call is hung up by the side that originated the call, the "h" routine is not extendable via a macro, or at least I have been unable to do it. My tests have included only SIP->SIP calls. If the originating side hangs up first: The macro is called from "exten =>

This is what I'm doing it gets you both sides of the phone call...small size...and playable on windows through a share. My notes: On redhat 9 I have to run the following command for asterisk to start LD_ASSUME_KERNEL=2.4.1 asterisk -vvvvgc [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${TIMESTAMP}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) ;exten =>

Hello Everyone, I am trying to change our /etc/sudoers (using visudo) to allow 2 commands to be run as root without a password, but it isn't working. Here is the part of the sudoers file that is in question. # User alias specification User_Alias FULLACCESS = doug, scott # members of the FULLACCESS User_Alias may run chown and chmod without a password FULLACCESS ALL = (root) NOPASSWD:

hi, Can anybody pls tell me, how to increase the time gap between 2 digits when you transfer a call. ie, the operator answers the call, and presses hash key to transfer, and then enters the extension number, some times, it timeouts too quickly before the operator enters the whole extension number (may be bcos the operator is slow). I tried the following, but it doesn't seems to be helping

Hi all. Thought you would all be interested to know that i, like many others, am experiencing issues with Samba and Photoshop 7.0. The problem is specific to Photoshop 7.0 when saving files to a Samba share via a W2k/XP OS. Users experience a disk full error when saving from Photoshop but have no issues copying/moving etc files to/from the share outside of PS. Well, after some testing, the

I have had this happen to me once: Encoding of a .wav file was interrupted somehow, so an incomplete .flac flie was generated. I accidentally ran a decode on that .flac file, which wiped out the original .wav file. I suggest that flac adopt one of the following to handle the situation that an output file is specified with -o or implied, and the output file already exists: 1. rename the

Hi everyone, I am trying to connect 2 asterisk servers via IAX, but it just fails to do so.. I'm using SIP to connect the IP phones on the LAN at the 2 different physical locations where each server resides and I'm able to communicate on my LAN via SIP without any issues. The problem is that I'm unable to make Asterisk servers talk with each other via IAX.. Here is my issue.

On this macro I keep getting exited non-zero on s,3, but s,3 is doing what it is suppose to do but the macro stops. Is there a way to make a macro ignore errors and continue to s,4? I have the lattes ver of sox 12.17.4. Also if I just run this line from the command line I don't get an error. [root@redhat monitor]# sox in.wav in-rev.wav reverse [root@redhat monitor]# [macro-record-cleanup]

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in

Dear All, I bought one IP PHONE from Integrated Networks which was showed to wiki too: http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks I have problems with the Asterisk authentication. It does't want to LOG IN to Asterisk; it always says "LOG ON FAILED". I'm using the IAX2 protocol and all paramters seems to be correct. Does somebody use the same IP PHONE with

hello i finally have success with winbind!!!! :) thanks everybody so, i use this acl's in squid external_acl_typqe NT_global/-group %LOGIN /usr/local/squid/libexec/wb_group acl FullAccess external NT_global_group "domain group" httpd allow FullAccess httpd deny all with this configuration all is DENID by squid. what???????? it's possible to write acl's specifics for

I know that Winamp has its loyal following. I've never used it much. Also, I'm a novice at computer music. I've always just bought CD's and never used computer music. So perhaps I'm an idiot or just not very music-software literate, but I so far haven't been impressed with Winamp, or the FLAC plug-in. Just for one example, I've got a dual-Althlon MP 2000 system

Hi, I'm running samba-3.0.14a_1,1 under FreeBSD 5.4-STABLE. I migrated all my data from samba 2 to samba 3, and I'm having a problem with some users. For example, user john can login perfectly to the network. It?s ok. But user smith not, him get the error ?The network name cannot be found?. But all configurations of john and smith are the same. I found this on Samba Official Hanbook:

I am working with the telantek.adsi file, and I was wondering how I would create a softkey for Transfer. I tried making a key definition and using SENDDTMF "#", but that didn't work. Is there another way I could do this? Also, does anybody have any ADSI scripts for use with Asterisk that they would like to share? Thank you for your time. __________________________________ Do you

Winamp3 is broken according to some posts here. Try Winamp 2.91 or the latest XMMS. Of course, XMMS requires the xmms-mp3 libraries. Try that and see if it helps. KJ <p>On Fri, 2003-09-12 at 15:56, Jeff Ousley wrote: > Thanks, all, for the pointers. I think I have > everything configured properly, but, something is > obviously still wrong. I'm using Kerry's guide with >

I just uploaded a new FLAC FAQ: http://flac.sourceforge.net/faq.html Let me know if I forgot any common questions. Josh __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com

I recently set up Asterisk with an Adtran TA750. All is well except the phones do not show the MWI. I have configured zapata.conf properly, as all phones will receive a stutter dial tone if there is a message waiting in it's assigned mailbox. Does anybody know how I might fix this problem? Thank you for your time __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free,

Hi list! Do anybody has success histories about using the Beronet app_saynumber application? For those that are hearing about for the first time, it?s a piece of software that enables the use of another language in say_number commands in asterisk dialplan or AGI scripts. Link to download: http://www.beronet.com/download/app_saynumber-beta-rc1.tar.gz I?m trying to compile it in