similar to: Asterisk On FreeBSD

Has anyone got this annoying sidecar to illuminate when users are on the phone? In my function key settings I have: Context: Active Type: Extension Number: <sip:4000@serve.address.com;user=phone> (4000 is the extension I want to see/dial on the key). I can press the key and it will dial the extension, it just won't illuminate when the user is on the phone or on DND Since I have

Without using Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ / \ User2 ------ Go to register ------- SIP2 ----- Whereis? --> DB / \ / User3 --

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of J. Oquendo > Sent: Thursday, April 26, 2007 6:47 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Asterisk brute force watcher (was FYI) > > Steve Totaro wrote: > > I suspect that

Hey all, after hooking up some Plantronics to some Snom's (3 320's 1 360), I noticed my client is having some form of feed back on the phone. Because of Snom's "inner oddities" this is how I got it to work. Plantronic --> RJ11 --> SnomHandset Port (on Snom Base) Handset --> Plantronic jack (bottom base in the front) If I placed Plantronic(RJ11) --> Snom's

Hello, i have call time irregularites in my asterisk CDR. I a currently using a mysqly backent to save CDR records and use this to generate bills at the end of each month. However, my users are complaining that they gety charged for even uncompleted calls (i.e. calls they make whaich have already be setup but canclled). i have noticed that only 'AWNSERED' and 'Busy' show up in my

Has anyone gotten this configured to show all extensions vertically instead of filling up the window. If so would you mind sharing your configuration Yes I have tried searching terms like +asternic +op_panel +vertical and a slew of others. Unsucessful though. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo echo @infiltrated|sed 's/^/sil/g;s/$/.net/g'

Long story short... Snom's ... Retrieve button... works when MWI is *NOT* lit but does *NOT* work when it is lit. Any advice Useragent : snom360/6.5.2 Function: F_RETRIEVE [root@pbx ~]# asterisk -rx "show version" Asterisk 1.2.13 built by root @ pbx on a i686 running Linux on 2006-11-17 16:35:22 UTC [gateway] exten => 201,hint,SIP/201 exten =>

So I have whose autoattendant is colliding with their extensions... Quick fix anyone? Second someone presses say a person's extension (101) ... Autoattendant sends them to the first context... [companyx-main-aa] exten => s,1,Background(companyx/companyx-main) exten => s,2,Background(silence/10) exten => s,3,Background(companyx/companyx-main) exten => s,4,Background(silence/10)

Hey all, I have a question for those administrating/building out systems with over 30 users on them. How often do you experience the dropped call phenomena. Would you care to share your experiences including what versions of * you were using, what kind of connectivity was present (T1, Fractional T, Intergrated T, DSL, Cable). Echo? Solutions? (e.g. we bought an X_Brand Echo Canceller). Also,

Hello All, I have a request from a prospectieve client to deploy a PBX capacity that can do up to 3000+ lines within a geographic region similar to a campus. The client wants analog lines for extensions and maybe VoIP for some backhaul traffic while the other traffic would be carrid via E1 channels. The client has other proposals to buy a mid, range telco switch from alcatel or simens but i am

Hi everyone, i am face with an asterisk use management interface, at the pressent, i am using AMP (asterisk Management Portal: http://coalescentsystems.ca/index.php?option=com_content&task=view&id=31&Itemid=57 ). Does anyone know a better and more documented management interface for * ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guys, I need your help ... I have a couple of DIDs that reach my Asterisk box .... But I'd like to set my DIDs automatically via Macro or other routine based on the number called by my agent ... Ex: My agent called 954-111-1111 ... So I'd like to set the callerid as 954-222-2222 (That is my DID) Thanks in advance, Marcelo

Victor Hoodicoff wrote: > > > I think your impressions of Aastra are outdated. Install the latest > firmware, download the latest documentation and test and THEN give an > opinion! Did you miss the part when I wrote I have Asstras sitting on my desk collecting dust. I program on average about 5 per month, deal with about 40+ per day. They're as impressive as that Hyundai in

So I have my extensions.conf (http://www.infiltrated.net/exten.stupidity.conf) shortened in case someone wants to look. Has someone encountered the following? I've racked my brain on this for too long... I have two contexts, day and night... Caller (Daytime) --> Dials an extension --> Caller hears extension ring on receiver --> Call goes through Caller (Night) --> Dials an

i am taking overr the administration of an existing production * PBX but i cant seem to find out which version of * this is. When i use the 'show version' coomandat the cli, i get this: Asterisk CVS-HEAD built by root@asterisk on a i686 running Linux on 2005-08-10 05:57:59 UTC can anyone tell me which version of * I am useing?or am i getting the command wrong -------------- next part

Hello, This is not a question directly related to asterisk. I am currently rinning ansterisk on a Debian server and i just upgraded my memory from 1GB to 2GB. However, my linux OS does not recognise the memory upgrade. The BIOS does, but the Debian Linux refuses to use the entier memory, currently, it registered only 900MB. Can anyone tell me why thi is and a solution to this?? My Debian version

Is plainvoip down? I've tried to contact them via email and their 800-956-3285; nobody is answering or replying to emails -- #Joseph

Has anyone gotten Asterisk to compile on Solaris 10? I have tried both 1.2and 1.4 and I get errors about editline. Actually it seems that 1.4 goes through more of the process, but thats not good enough.... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061116/82464f01/attachment.htm

I've dug down as far as I could on www.uspto.gov for anything remotely close to what is going on with Verizon and all searches end with only two possibilities in regards to what is going on. So unless the patent was issued to someone else and Verizon bought it, these are the only two possible patents this case could be based on... US 7,142,646 B2 Voice mail integration with instant

Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16