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Hi all! I have trouble with connection to AMI 1.1 wich enabled on Elastix "*Asterisk Call Manager/1.1* *Action: Login Username: admin Secret: qweasd123* *Response: Error* *Message: Missing action in request*" Elastix versions: "* Kernel* * Linux(x86_64)-2.6.18-348.1.1.el5* * Elastix* * elastix-2.4.0-1* * elastix-portknock-0.0.1-0* * elastix-agenda-2.4.0-1* *

We have a problem in that calls are dropped after 15 minutes (on both internal and out going calls, incoming calls do not seem to have that limit) How do we fix it? This is the version on that PBX Kernel Linux(x86_64)-2.6.18-371.1.2.el5 Elastix elastix-2.4.0-8 elastix-a2billing-1.9.4-5 elastix-addons-2.4.0-10 elastix-agenda-2.4.0-14 elastix-asterisk-sounds-1.2.3-1

Hi folks, I've been a trixbox user for a few years now but I'm thinking about jumping ship. Trixbox is great, but it's missing two features out of the box which are really important to me: outbound faxing (hylafax) and imap voicemail. I see no indication that they will be included anytime in the near future, so I have a choice to make - I've looked around at Elastix,

Hello Folks; I have an Asterisk server Asterisk 11.7.0 built by root @xxxxxxxxxxx on a x86_64 running Linux on 2013-12-27 18:47:44 UTC No FreePBX, no AsteriskNOW, no Elastix. Just Asterisk. Is there an API out there that anyone knows of that I can pass commands, etc to Asterisk? Creating Extensions, adding voicemail users, setting up voicemail, etc? I'm kind of clueless. Is there something

> I'm very new in Asterisk and VoIP, and of course I have a problem... :) > > Well, my problem is, that Deutsche Telekom wants me to change my ISDN > to VoIP... :( > I must do that, since I have no alternative. > > Well, I have now two VoIP-phones (Thomson ST2022 and KE1020A). I can > configure my two numbers by Deutsche Telekom and I got now an extra > number

Dear community, Please help. I've been looking around the internet (and in this great forum) for help with DUNDi setup between servers (I'm using Elastix) and while I can get my servers to lookup extensions on each other very well, I have not been able to successfully make calls between servers. For my test environment, I have 3 servers setup for now, and these are the steps I've

I'm impressed with the call log reporting that Switchvox offers. There's a screenshot here: http://www.switchvox.com/sv?cmd=screenshots&pic=13 Can normal Asterisk do this? I installed AsteriskNow on an old computer, but I couldn't find anything like that in the GUI. I found the CSV file in /var/log/asterisk/cdr-csv, but that's not much help unless there's some other

Hello, I need to have an extension on a SwitchVox server dial out to one on an Asterisk (FreePBX actually) box which will host a voice directory. The Asterisk server will then need to dial one of the SwitchVox extensions if it gets a voice match. Anyone has done that, and could let me know how? So far it looks like IAX peering (what SW calls "SwitchVox peering") could work? Thanks in

I just inherited a client that is using a Switchvox system. I normally install a CentOS based system with freePBX and some custom endpoint management stuff for Polycom phones. This Switchvox is making me feel a bit stifled. I am having nightmares of another recent encounter with Trixbox Pro. Can I really not ssh into this box? If I could is there anything useful that I might change

Hello, My first attempt to get dahdi running on 1.4.28... with a Rhino 8 port modular card and a single FXS module. Got the Rhino card installed and the machine sees it: root at pbx:/etc/dahdi# dmesg | grep rcbfx [ 71.985309] rcbfx 0000:04:00.0: PCI INT A -> GSI 21 (level, low) -> IRQ 21 [ 71.985440] rcbfx 1: Rhino PCI BAR0 50100000 IOMem mapped at ffffc90008d7c000 [ 71.985504]

Hi, I currently have some systems on AsteriskNOW 1.7 & have been happy with its clean simplicity & reliability. Are many people here using AsteriskNOW 2.0.x? How do you feel about it? Did Digium stick with their previous philosophy of keeping everything very vanilla & making it clean & simple for someone who understands how to manage CentOS, FreePBX, tftp, ntpd, etc. but

Hello, Is it possible to set things such as parts of config files are edited though AsteriskNOW GUI while other parts remain "hand editable" ? AsteriskNOW website include screenshots but not much information (such as user manual) beside that. This thing was the one that kept us from using freepbx (let me say I don't mean it's not possible with freepbx : I mean we couldn't

Hi All; Based on what I have to use Trixbox or FreePBX? Can someone advise? Regards Bilal

Hi list, Anyone knows about the date of the official (stable) release (v1.0) of AsteriskNOW??? It's supposed to be at the end of this year, which is very close now with no signs of it. Thanks... Raul -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071218/c543b407/attachment.htm

I am currently using AsteriskNow v2. What I am trying to accomplish is having all calls from an area code go directly to the person responsible for that area. While searching for a solution for this I did come across a post that had a few examples. So Josh at extension 1902 would receive all calls from the 808 area code. exten => s,1,GotoIf($${CALLERIDNUM:0:3}" = "808?1902|1)

I am having a very tough time trying to replace an Elastix 2.X install running as a virtual machine on ESXI 4. I tried using the Freepbx 14 ISO that installs CentOS 6 along with Asterisk 13.16 but I keep getting random segfaults: [175711.476685] asterisk[2942]: segfault at 188 ip 00007fc6c41abffc sp 00007fc608575890 error 4 in libasteriskpj.so.2[7fc6c4144000+14c000] I then proceeded

Hello all. As part of a project I'm working on to migrate from Asterisk 1.0.11.1 to the latest LTS version, I'm looking into providing a HA/failover solution for the new Asterisk installation I'll be deploying. It would appear my best bet would be to use the R850 Digium appliance. Does anyone have any experience running one of these devices along with AsteriskNOW? I ask because

Hello all. I was hoping someone out there might have some advice or suggestions regarding an upgrade from an archaic Asterisk version. I've been given the daunting task of upgrading a very old Asterisk-1.0.x install to a recent LTS version. I'll also need the install to have high-availability and failover support. From my research, it would appear that AsteriskNOW-3.0 might be my

I have been asked by my church to recommend a VoIP system which can do the following. They do internet radio shows which are sometimes broadcast on radio. They are looking for a system which does the following for about 5 agents, exactly as they have described it. 1. Take incoming calls 2. Put them on hold if there is no one to handle the call immediately, or transfer them to an available