similar to: #Transfer - Timeout is configurable?

Say I have agents using a softphone like eyebeam that has 6 lines. They log in to the queue. Say there are 3 agents in my queue. 3 calls come in and all three agents are on a call. Now a fourth call comes in. Is it possible to have it setup so that the 4 call rings on line 2 of one of my agents, if they don't get it within the time limit it rings on line 2 of another agent and so on. An

Hello, I''m running mongrel_cluster and have been using it successfully for quite a while now. However, today I ran a "cap deploy" and started seeing the following error: !!! PID file /var/www/apps/jco/shared/log/mongrel.8013.pid does not exist. Not running? for each of the mongrels that should start up. I''ve googled for solutions but haven''t found any

---------- Forwarded message ---------- From: Marco Mouta <marco.mouta@gmail.com> Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: asterisk-users-request@lists.digium.com Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me

Hello I have been working on a little window dialog script - please see <http://net4visions.com/dev/window/window.htm> http://net4visions.com/dev/window/window.htm for a preview. the necessary files can be downloaded here: <http://net4visions.com/dev/window/window.zip> http://net4visions.com/dev/window/window.zip .I''ve included just the basics - please change the path in

Just today I received a weird e-mail that contained a worm (WORM_GREW.A) attached to it. From : Vinoth Kumar <vinoth@vinjey.com> Sent : Tuesday, January 17, 2006 6:29 PM To : <ivo_mmm@hotmail.com> Subject : Fw: Funny :) Attachment : Attachments001.BHX (0.13 MB) MIME-Version: 1.0 Received: from vinoth ([59.144.59.103]) by bay0-mc4-f5.bay0.hotmail.com with Microsoft

Hi all Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: http://bugs.digium.com/view.php?id=5884 Haven't tried it out yet. Benoit Panizzon -- I m p r o W a r e A G - System Services ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz

Does anyone know how to remove the browse list comments listing? In other words, in our instance, when you browse the my network places, both the resource name and comments are lengthy Samba 3.0.13. listings. Thanks,

I have been scouring this list''s archives and have consumed any and all how-to guides I could find but am still unsure. All discussion in this list on this issue is around hvm/vti DomU''s. Has anyone been able to attach a vnc client to a vanilla (non-hvm) DomU using the VNC console setting? On an FC5 (2.6.16-1.2080_FC5xen0) kernel deployed on vanilla (non-hvm) hardware,

Boa tarde, Ap?s alguma experi?ncia com o Asterisk, e com muito ainda para aprender, gostaria de saber se h? algu?m nesta mailing list que pretenda criar um Asterisk Users Group para Portugal. Visto que acaba sempre por ser uma enorme aprendizagem ( valor acrescentado) a partilha de experi?ncias/problemas e solu??es nas implementa??es Asterisk. H? spre detalhes que variam entre os Telco's de

Hi all, I've the current scenario: User "A" - Zaptel call incoming in my Asterisk to my SIP user "B". "B" gets the Call. "A" says : "B" i would like to call PSTN user "C" "B" places a call to user "C" and asks if "C" wants the call from "A". "C" says yes i want, then B needs to

Hi guys, I've setup on box with a TE110P and time to time I need to access remote equipment outside of our office and use a data channel. I'm wondering if do I need to buy a POTS line only for this time to time acess or what's the easiest way to do that via my TE110P on asterisk box. I know that is possible data transmission with this Digium Card, I'm wondering how... Any tip any

Hi all, I've a Planet VIP-150T VoIP Hardphone connected to Asterisk. When I'm in a call and i press Hold button, the other party starts listening Music on Hold but then when i press the button again to get the call back it doesn't work! I've checked asterisk CLI: -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 --

Hi all I'm deploying a VoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions. On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application. I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX

Hi I'm very new to Asterisk and this is my first posting to this mailing list. I got a Asterisk@home V2.8 working, and now I'm trying to use Asterisk.NET (http://sourceforge.net/projects/asterisk-dotnet) to get call events to my C# program. Asterisk.NET comes with a sample program called Asterisk.NET.Test and it uses the following default constants for login: const

JABBER: gtalk_account OUTGOING: <?xml version='1.0'?><stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to=' gmail.com' version='1.0'> localhost*CLI> jabber show tes JABBER: gtalk_account INCOMING: <?xml version="1.0" encoding="UTF-8"?><stream:stream from="gmail.com"

Hi all, I've my Server running well, then sometimes Sjphones looses registry and it only works well again if i restart the pc running sjphone. Has any one experience this? Best regards, Marco Mouta

Hi all, I've 2 * tryning to connect each other Server A is already registred on server B But server B never registers in server A I always get this: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ Timestamp: 00018ms SCall: 00004 DCall: 00003 [XXX.XXX.XXX.XX:4569] CAUSE : Registration Refused CAUSE CODE : 29 Any tip? Best regards, Marco Mouta

HI all, My asterisk for all my users, everything was fine for 3 days, but now i can't access it. But it is running... Could any one help me on this? Best regards, Marco Mouta

[macro-hangupcall] exten => s,1,ResetCDR(w) exten => s,2,NoCDR() exten => s,3,Wait(5) exten => s,4,Hangup Hi all, currently i've been getting troubles with SIpphone Sjphone and Xlite seems also to get delay but no crash on hanging. I found that Ast@home is executing this Wait(5) and it seems to me that Sjphone is giving timeout error because of it... Why is this 5 seconnds? any

Hi guys, I've been installing Asterisk 1.4 with Asterisk addons, and i could notice that in /usr/lib/asterisk/modules/ doesn't have cdr_addon_mysql.so even after compiling Asterisk Addons! In fact the cdr_addon_mysql.c exists, but it doesn't seems to be compile when i run Asterisk-Addons: make && make install Any one can help me on this? -- best regards Marco Mouta