similar to: Re: Can't get Polycom auto-answer to work (Solved)

Displaying 20 results from an estimated 1000 matches similar to: "Re: Can't get Polycom auto-answer to work (Solved)"

2005 Mar 10
1
OT: AstLinux 0.2.2 released
Hello Everyone, I have released AstLinux 0.2.2. There are way too many improvements to list here, but here is a short summary: Linux 2.4.27, iptables, mini_httpd (with PHP & SSL), phpconfig, AstShape traffic shaping, tftp server, OpenSSH, proftpd, Soekris Net4801 and Pentium-MMX and higher x86 support. There is actually WAY more software, but I couldn't possibly list it all. It
2008 Sep 26
2
Old 16bit windows game in Linux!!!!
I have Ubuntu 8.04 and I am trying to run and old Windows 95/3.1 MPOG game called Legends Of Kesmai. The Game needs to be in the root directory i.e. C:\legends and also needs to be run in 16 bit color. I have managed to change my color to 16 bit via editing my xorg.conf and changing the 24's to 16's. However when I try to execute the lokclient.exe file, it just closes itself without any
2005 Jan 06
1
Re: Asterisk-Users Digest, Vol 6, Issue 73
Hi John, Kevin, Tor and Wiley (and everyone else) - >> I guess the phone just doesn't register as busy when there is only one >> call on a line. It has to have two calls on a line appearance to >> register as busy. Has anyone figured out how to disable this hold >> feature and just have the second call go to the second line, the third >> call to the third line,
2005 Jan 27
0
Re: Polycom and call waiting again...
>Message: 10 >Date: Wed, 26 Jan 2005 17:53:39 -0500 (EST) >From: "Sean A. Newton" <nester-asterisk@wewt.net> >Subject: Re: [Asterisk-Users] Re: Polycom and call waiting again.. >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: >
2005 Jan 06
0
FW: Re: Polycom IP500 - problems with multiplesimultaneous calls
Adam, Tor sent this one a little while ago that looks really promising for solving the problem. Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tor Setane Sent: Thursday, January 06, 2005 2:09 AM To: Noah Miller Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re:
2005 Mar 14
2
Has anybody experience with SetGroup / CheckGroup commands?
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
2005 Jan 24
1
SetGroup and CheckGroup problems
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
2004 Aug 11
1
limit incoming calls to sip extens
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
2004 Oct 01
1
Agent Login Problems
See comments below. Henry Devito wrote: > Here's the problem. When I call 555 to login, it asks for the agent ID > which I enter as 501, it asks for the password which I enter as 1234, > then it asks for the extension I dial 501 It then says that extension is > not valid. What am I missing? Of course 501 is valid I can make and > take calls from it now. > > >
2005 Mar 24
0
Re: IP-500 config
Hi Noah - > I got everything to load via ftp. The phone appears to correctly boot > from the config files. I also put the latest firmware there and the > phone sucessfully loaded it. > > For some reason, the phone and * don't see each other. This is the > part > that confuses me. Any clues as to why the phone won't register? It's not often I get to
2005 Jan 28
2
Polycom changing policy - allowing firmware downloads?
I don't know if Polycom decided to change their policy or not, but I just went to their website and when you click on the downlooad link, it will now present you with an option to join their extranet. I did that and I got the following message immediately: <snip> Access to certain Resource Center sections and content are subject to approval. Eligible members, will receive further
2004 Aug 31
2
limit the length of extensions
How do I limit the length of an extension? In my test IVR/Automated Attendant (whatever it's called), at the beginning it plays "if you know your parties 3 digit extension, you may enter it now) and then it gives a list of options. If the caller puts the 3 digit extension, it goes through fine, if they press 1, or 2 it goes to the selected menu option, but if they dial 91235551212 it
2004 Jun 25
1
Howto: Use setgroup, checkgroup to check incoming and outgoing client limits
Hi there, I was wondering how I can use setgroup and checkgroup for perfom incoming and outgoing limitation checks. I've have some users that doesn't what to be able to recieve more than 1 call at a time, and I also want to limit a users outgoing call abilities. Any help would be greatly appreciated. Kind regards Cf --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus
2004 Jun 28
1
SetGroup and CheckGroup
Does anyone know if SetGroup and CheckGroup apply to only current context or is it per server based? Ta SJ
2005 Feb 23
0
IAX Trunking capacity enforcement
Hello, I am trying to come up with a good way to enforce a limit on the number of simultaneous calls that can occupy an IAX trunk at any given time. I have searched around and so far can't locate a config option that would directly label a IAX trunk with a specific number to obey (is there one?). Based on examples for the SetGroup and CheckGroup commands, I am thinking of using SetGroup
2006 May 04
0
SetGroup and CheckGroup. Need some help on the dialplan
>From this list I found that I could use SetGroup and CheckGroup to do what I wanted. But I'm not quite sure how I do it. The case is that I have 3 user groups, and one main group. The main group is for all the incoming calls from external phones. The main group should be allowed to have 3 calls at the time. The 3 user groups are internal groups that I want to limit by ony having one
2005 Jan 04
0
Does congestion exit on a different priority?
Customer is having problems with his internet connection, I have in my context: [jimballboutiques] . exten => 1235690251,1,SetGroup(customer) exten => 1235690251,2,CheckGroup(3) exten => 1235690251,3,Dial(SIP/jimball,20,r) exten => 1235690251,4,VoiceMail(u1235690251@jimballboutiques) exten => 1235690251,103,VoiceMail(u1235690251@jimballboutiques) . Now I've had it
2004 Jul 30
1
FW: Limit incoming calls to SIP Channels
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
2005 May 10
1
Limiting outbound calls
My VoIP provider allows me to have more than one call outbound on the same line simultaneously, for some reason. I am pretty sure that they do not want this to happen, so I'd like instead to limit each line to one call. I do not want the users to have to dial another prefix to go out on another line. Is there any way to add multiple accounts for my _9. extension and have Asterisk
2005 Jun 19
2
outgoing call routing
I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip extensions and a regular phone connected to the box. All routing works fine from the regular phone connected to the box, whether its going to FWD, broadvoice or the PSTN. The problem I am experiencing comes from making calls from the sip phones. They get routed correctly to the sip and iax trunks but when making calls