similar to: SV: SV: Queues and call waiting indication

Hi, This issue has been discussed probably a million times on every asterisk forum in the world and I have the same problem too. Another problem you would have with the agents is that when they make an outgoing call they are not regarded as "busy" by asterisk and it sends more calls to the agent if it has call waiting enabled. This behaviour is totally senseless since the whole purouse

Hi, I'm running 1.2 beta1 in a mini call center. I have 3 queues with 10 operators, and I'm running into some trouble because when all the operators are busy answering call asterisk still sends them more, resulting in a "beep beep" (call waiting) over and over again in Xlite audio. An easy solution woud be the use of a "single line" user agent, like firefly, still

What kind of help do you need then? Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Zach A Skickat: den 6 februari 2006 18:31 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' ?mne: RE: [Asterisk-Users] Help on queues There is no good help on wiki and voip-info.org, I've

Thanks! Do you have any suggestions on what I might do next. I have the phones, I have asterisk, and I have everything setup. But i can't get the login to work with the Polycom function. Nothing happens...and I can't find any readmes' or manuals. Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com

OK! I'm sorry about this. As I wrote earlier I'm totally lost... but I will try to explain the problem in steps bellow, ok. 1. I installed the rpm's for samba, Version 2.0.2a-ssl I think this is the version distributed with redhat linux 7.0 2. Then I changed the parameters in the /etc/samba/smb.conf file, and in this file I added the folowing parameters. [global] netbios name

Thanks for the quick response! I read about logrotate at voip-info.org but I didn't quite understand it. I'm no asterisk/linux expert unfortunately. First of all. What exactly does happen when I run: /usr/sbin/asterisk -rx 'logger rotate' Does it clear the file and create a new one? Can I run this manually without any interruption in the system? And what does the script do? I

The update worked like a charm! Hold music is as cheesy as ever! Thanks much, this list is a life saver! Dan ------------------------------ Message: 2 Date: Fri, 18 Mar 2005 09:16:59 -0600 From: Eric Wieling <eric@fnords.org> Subject: Re: [Asterisk-Users] Redhat 9 Music on hold To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>

Any expert suggestions most welcome , I am really stuck .. thnx -----Ursprungligt meddelande----- Fr?n: Ashish Garg Skickat: to 2003-05-08 09:04 Till: Jair; John H Terpstra Kopia: samba@lists.samba.org ?mne: Samba Installation help for a domain HI Guys, I need some help regarding Samba Installation for a domain. My Windows 2000 clients connect to a paricular doamin. My Samba

The diagnostis.txt file don't seem to solve this problem. Now my samba server is now available for browsing, I dont know how I should solve this actually I'm totally lost I even tried to erase the rpm file and compiled a source file. Mvh / Best regards Daniel Andersson ------------------------------------------------------------ REJLERS INGENJ?RER AB R?dhusgatan 15, S-541 30 SK?VDE,

This is what i found on Cisco's site: "Symptoms: Media negotiation fails for SIP calls and the terminating gateway replies with a "488" message to an Invite message. Conditions: This symptom is observed on a Cisco platform when the terminating gateway is configured with the G279B (annex B) codec and when the Session Description Protocol (SDP) for the incoming Invite message

I tried with following: require ["imap4flags"]; if not hasflag :is "\\Seen" { setflag "\\Seen"; } And then this in plugins.conf: plugin { sieve_plugins = sieve_imapsieve imapsieve_mailbox1_name = Trash imapsieve_mailbox1_before = file:/etc/dovecot/sieve/trash.sieve } It works in outlook, the mail is opened (mark as read) when it goes to trash. But in

Hi Alessio Yes, the problems you report do seem similar to the issues I had. I found on the Dells that the audio prompts were very choppy and played slower than normal. Occasionally there would be 'bursts' oav a second or so of 'good' audio. I also suspected IRQ issues but the Dell Mobos had no way of adjusting them. Best thing is to try and get the card on its own unshared

It's exactly the same. Except that I use tdbsam instead of ldap and the error message therefore also is different in the log file. But the example and result is the same. Do you have any idea of workaround or fix? Cheers, Joel -----Ursprungligt meddelande----- Fr?n: Doug Campbell [mailto:doug@bpta.net] Skickat: den 11 maj 2005 10:49 Till: Joel Larsson, PF, Posten; samba@lists.samba.org

I know it is not dovecot who should fix this. But anyone using dovecot is using an MTA, and receiving spam ;) I know how to look at email headers. Spf and dkim is not solving anything here. -----Original Message----- From: Sebastian Nielsen [mailto:sebastian at sebbe.eu] Sent: donderdag 11 juni 2020 10:23 To: Marc Roos; 'dovecot'; 'users' Subject: SV: handling spam from

Sorry about that, its just outlook that does that by default. But manually deleted your adress now in reply. I don't know what you mean with "top posting"? What I mean is that if you have another security on the connection (be it physical security - the connection doesn't go over public means, or VPN - connection level encryption) then you don't need another encryption on

Hi, I would like to enable mysql handling of voicemail boxes ... following that tutorial http://www.voip-info.org/wiki-Asterisk+voicemail+database so I modified the makefile of /apps directory to include USE_MYSQL_VM_INTERFACE=1 and copied mysql-vm-routines.h in the /apps dir, set up the db and some boxes in the table, also edited the voicemail.conf file. Everything compiles just fine, then

Hi, I'm testing realtime right now, it does not seem to me that realtime contexts can be included in normal context, like this [sip] include=>sip-dial exten=>i,1,Hangup [sip-dial] switch=>Realtime/sip-dial Am I getting it wrong ? Tnx ! -- Best regards, Alessio mailto:afoc@interconnessioni.it

Hi, I get this message after password request in voicemail app: Unable to create lock file: No such file or directory Anyone got a clue about fixing that problem ? I can't understand what directory or file we are talking about .. Tnx for any help! -- Best regards, Alessio mailto:afoc@interconnessioni.it

Ok, I'll post all the messages on the screen. -----Ursprungligt meddelande----- Fr?n: H. Peter Anvin [mailto:hpa at zytor.com] Skickat: fr 2003-05-30 22:21 Till: Olsson Lars Kopia: syslinux at zytor.com ?mne: Re: [syslinux] Problems with PXELINUX 2.05pre1 Olsson Lars wrote: > Reading config based on hw address is really what we need, so we quickly > downloaded

Hi I'm running exactly the same setup. Asterisk is running on my FW/NAT/Router with two interfaces. My local phones are situated behind the NAT and connects to the outer interface of the */FW/NAT/Router. * is then connected to my SIP providers (since I'm only using the SIP-part of *, PSTN connection through my SIP-provider). Works fine! rgds, /staffan kerker sweden -----Ursprungligt