similar to: Aterisk App ICES Question

Hello, Since outgoinglimit is EOL'd, I've implemented SetGroup/GetGroupCount to ensure that SIP clients will only have a single call at any time. Works perfectly for simple calls using Dial(). I'm now struggling to find a way to similarily limit 2nd calls to SIP clients that are Agents, who receive their calls from a Queue(). Is there any way to accomplish this (without writing

Hi all, I am just scratching my head trying to work out a way to use SetGroup to check busy status on a sip to sip call. The complication is that one call can't be in two groups so I have got no way of setting busy status on both the calling and called party. Has anyone got a way around this. Thanks Daniel -------------- next part -------------- An HTML attachment was

I am checking on the SetGroup / CheckGroup commands, but I have some troubles to undestand the examples. SetGroup(moh) can be moh anything as I like? Usually moh stands for "music on hold" CheckGroup(1) checks if somebody in in group "moh". Does it mean I can only have one SetGroup(xxx) ?? When I look at example 2 than I see two SetGroup commands and one CheckGroup

I have a rather long dial plan, but it includes support for call waiting. However, the setgroup checkgroup commands don't seem to be working. Can anyone help on this one? Excerpts are below. First exten-vm is dialed and then dial-new. As I understand, priority 1 increments the active channels for the caller and then in "dial-new" priority 8 increments for Arg3, or the Callee

Hi All, Can someone please tell me how to limit incoming calls to SIP channels using the SetGroup & Checkgroup command. I don't want any call waiting on SIP channels and you are somehow meant to be able to do it with these commands. Many Thanks Daniel Niasoff

Hi All, We have just installed Asterisk@Home. It was straight forward as promised. However, I cannot find any guides or tutorials on how to administer this version of asterisk. We plan to install a bunch of Cisco 7960 and 7905 IP phones. I have a test phone which has already been upgraded to SIP 7. Now the box is ready but we don't know what the next step is!! Any help is appreciated.

Hi iam working with asterisk with mysql Realtime when i have confgured and run the asterisk iam getting the following error i dig all the places for help could not find the results could some one help me what is wrong iam using 1.2.5 on FC4 Mar 20 23:04:52 NOTICE[2054] cdr.c: CDR simple logging enabled. Mar 20 23:04:52 NOTICE[2054] indications.c: Removed default indication country

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

Anybody knows why ARI gives this error message when I enter extension number and password. *Warning*: file(/var/spool/asterisk/voicemail/default/222/INBOX/msg0000.txt): failed to open stream: Permission denied in * /var/www/html/recordings/modules/voicemail.module* on line *525* It doesn't show the voicemails, although it shows that there is 1 or 2 voicemails in the INBOX. -- Zeeshan A

hi, i have strange problem with call-limit/groupcount limiting. i set up limit of 2 calls. i'm using both methods but a for few times i have problem with successfull fraud with more calls than 2 asterisk is 1.8.22 someone with the same problem? any ideas how to solve or debug this problem? -- --------------------------------------- Marek =======================================

Dear all i'm planning an upgrade of some asterisk installation from 1.4.32 to 1.6.0.28 (as i think it should be the most stable now). Reading the UPGRADE-1.6.txt file i've noticed that: * SIP: The "call-limit" option is marked as deprecated. It still works in this version of Asterisk, but will be removed in the following version. Please use the groupcount functions in the

hi, i'm using groupcount to limit max calls in pbx i want show/graph made calls usage it's possible make this from cli/ami? something like asterisk>group show usage name channels group1 3 group2 5 google doesnt help thanks --------------------------------------- Marek Cervenka =======================================

I've been trying to piece my way through the encoder_example.c program to better understand how to encode files as ogg/vorbis. I'm stuck on two sections of the code. This is the first /* uninterleave samples */ for(i=0;i<bytes/4;i++){ buffer[0][i]=((readbuffer[i*4+1]<<8)| (0x00ff&(int)readbuffer[i*4]))/32768.f;

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

I have my icecast and ices working nicely together and it sounds great. The issue that I have at the moment is that the ices stream dies after about 18:00 CPU hours. Is this a common problem? Anyone know any tweaks so that I can keep it running? I run FreeBSD 4.10 Rob

Hi all, I've been using the following method to limit calls to sip clients to 1: exten => 200,1,SetGroup(200) exten => 200,2,CheckGroup(1) exten => 200,3,Dial(SIP/200) exten => 200,103,Busy This works fine for a single extension. However, I also need to dial groups of sip clients. It appears that SetGroup can only be used once per channel. This (useless) example would not

On Sunday 28 July 2002 02:06 am, you wrote: > On Sunday 28 July 2002 01:30 am, you wrote: > > Whenever I try to run it (ices)I get errors: > > > > cadmium:/work/dl/Audio/streaming/ices-0.2.3 # /usr/local/icecast/bin/ices > > --help /usr/local/icecast/bin/ices: error while loading shared libraries: > > libgcc_s.so.1: cannot open shared object file: No such file or

On Mon, 2003-11-24 at 21:58, iceuse@kezako.net wrote: > Hello, > I changed my config for ices and added passthru. Well that will reduce CPU a lot, as the ogg streams are only rebuilt, no re-encoding is going on. > I will see how it will run now. > I have realtime I think: realtime needs privileges to be enabled, and as the log shows the

On Sat, 2003-11-22 at 10:00, iceuse@kezako.net wrote: > Hello, > I had a very strange behavior of ices: > [2003-11-22 10:27:17] EROR playlist-builtin/write_ogg_data failed buffer allocation This would indicate that too much audio was in flight to the output, if it was running ok for some time then this probably means that the system became very busy preventing the runner from freeing up

You are trying to use ices 2 with an ices version 1 config file. On Friday, 13 June 2003 at 08:01, admin wrote: > OS: FreeBSD4.8 > > okay I installed ices from /usr/ports/audio/ices > > copied ices.conf.dist to config.xml > > can somebody explain to me why the configuration file is not getting loaded > > root@hurricane# ices