search for: u100s

Hi Everyone, We are using two Sangoma U100 (USB FXO) units connected to an Acer Aspire Revo (little PC running on Atom). The units work beautifully except for Monday :-) It maybe a conincedence or maybe the fact that Saturday/Sunday is off and something happens where one of these U100 modules goes into sleep and that's when all the 4 Dahdi channels are lost. So, I have been getting Monday

Hi, I recently purchased an MSI Wind Netbook, just slightly bigger than an EeePC, and IMHO just one step above the mere toy category. It's got a 10" monitor, 80 GB SATA HD, and the keyboard is just big enough so I can type with ten fingers (with a little exercise). The thing came preinstalled with Windows XP, and I've spent the last two days figuring out how I could possibly

I am looking to build a small PBX for an office that has 3 incoming analog lines and less than 10 extensions. For the Asterisk server I am going to use a small form factor PC with no-PCI slots so the FXO interface needs to be either FXO->SIP or USB. Can anyone make suggestions? I am looking at an AudioCodes MP114 FXO or possibly two Sangoma U100's but don't have experience with

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

I have one analog line hooked in my Asterisk box using an x100p (I think that's the model number). When I do this in my extensions.conf: exten => 1200,1,playback(pls-wait-connect-call) exten => 1200,2,Dial(Zap/1/5555551212,20,rTt) exten => 1200,3,VoiceMail(u100@lightwavetech.com) exten => 1200,4,Goto,t|1 The phone rings beyond the 20 second timeout and never really goes to the *

hello, I''m running XEN 2.0 on IBM ThinkPad T23. Now the weird thing is that I get two different outputs from /sbin/lspci depending on whether I run 2.6.8.1-xen0 or 2.6.8.1-bproc. In particular the output from 2.6.8.1-xen0 seems to be missing those 4 lines 0000:00:01.0 PCI bridge: Intel Corp. 82830 830 Chipset AGP Bridge (rev 02) 0000:00:1e.0 PCI bridge: Intel Corp. 82801BAM/CAM PCI

I know I've seen this reported already, and I can't remember the fix. I have two ATA186s talking to an asterisk server. When I call in on an outside line, both ring, and I can pick up either and talk. But if I try to call from one of them to the other, the remote end rings just fine in both cases, but then as soon as asterisk bridges the two channels, the remote end sends a

hi, i am trying to do autoattendant but failing. as in the manual i inserted the background(welcome-mainmenu) file so that after the sound the caller can dial the extension he wants to call. i figured that the background sound wasn't coming in the asterisk. how do we do this without first loading the welcome message? for example after certain rings the caller can dial the extension no to

hello i'm on debian wine 1.3.37 and i try to use iriverplus4 for an iriver U100 i'm french and my english is not very good sorry. papa at debian:~/.wine/drive_c/windows/system32$ wine iriverplus4.exe err:ole:CoGetClassObject class {25baad81-3560-11d3-8471-00c04f79dbc0} not registered err:ole:CoGetClassObject no class object {25baad81-3560-11d3-8471-00c04f79dbc0} could be created for

I have an issue getting any incomming calls When the phone rings something picks up and gives it a fast busy. There is no one using Zap/2 it does the same thing with voicemail and voicemail 2 you can see the console output below, I would love any help anyone could shead on this issue, Michael NOTICE[1192484144]: File chan_zap.c, Line 4270 (ss_thread): Got event 2 (Ring/Answered)... --

I have a small * system in my home (1 U100S, 1 X100P, 1 BT101, and 1 SPA2000) to handle my requirements. I would like to add Music on Hold and have been watching the forum to see if something would come across on this topic. The difference I am interested in is getting the music from a radio or someother external source. All references to...

Due to changes at the office, I'm finally getting around to setting up an AA to deal with incoming calls. One of the big changes is that we're dropping the old alphanumeric pager and will just send pages to our phones. I've got the outbound greeting message working in a test context no problem right now, but I'm kind of stuck on how to capture a DTMF sequence from a user and

Hi *, I have set up a very simple asterisk configuration where I intend to be redirected to the voicemail whenever I dial 100 with my kphone SIP client. The problem is that asterisk can not find the stream 'vm-theperson'. I have made a non-standard installation (since I am just testing), and that file is located in /mnt/tr2/fake_root/installed/usr/local/var/lib/asterisk/sounds. 1. How

Hi, I have a weird situation where I have a noop command putting the callerid of the caller on my asterisk console so I know who is calling as a test, but it is putting the callerid of my extension in instead of the callerid of the incoming line. My /etc/asterisk/zapata.conf is [channels] context=default ;switchtype=national usecallerid=yes cidsignalling=v23 cidstart=polarity hidecallerid=no

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I don't know if this is the right list, but here goes what happened recently to me. I have an IDE disk (hda: QUANTUM FIREBALLP AS20, ATA DISK drive / hda: 39851760 sectors (20404 MB) w/1902KiB Cache, CHS=2480/255/63, UDMA(100)) connected to 00:1f.1 IDE interface: Intel Corp. 82801BA IDE U100 (rev 12) I've decided to make a

I have a simple dial plan to cascade calls when the first phone does not answer: exten => 100,1,Dial(SIP/1000,10,tr) exten => 100,2,Dial(SIP/1000&SIP/1001,10,tr) exten => 100,3,Dial(SIP/1000&SIP/1001&SIP/1002,10,tr) exten => 100,4,Voicemail(u100) Problem is that the once the call goes onto the second and subsequent steps exten 1000 cannot answer the call. When the user

Can anyone recommend some phone sets that are adsi compliant and work well with asterisk?

I am using the following macro to dial a ZAP channel. When I dial in, * answers and I go to voicemail. I never hear any ringing, though. It doesn't work with the Ringing command before or after the Dial command. [macro-zapdial] ; ; call a ZAP extension for ${ARG2} seconds, and then voice mail ; ${ARG1} - Extension ; ${ARG2} - Time to ring exten => s,1,Dial(ZAP/${ARG1},${ARG2}) exten

Hi, I have a deltathree account and I can place calls but I can't receive calls. I use Grandstram sip phones. When I call my deltathree phone # the voicemail is answer :(( I need some help and solutions from the guys who allready are using deltathree. I search on Internet and I try all types of configurations... :( This is my configurations files: - sip.conf - [general] port = 5060