similar to: Scaleable Solution for small office

Alright, this may seem like something relatively easy to do but I must be missing something or had a neuron misfire. I am trying to get The Status lights on my Snom200 hardphones to display the status of each one of my PSTN lines in my Asterisk server. Current Config: 3 X100P cards Asterisk CVS-02/25/04-18:06:52 5 Snom200 phones I am currently using the following macro to dial out

Have a question about implementing Call Rollover with my current extensions.conf configuration. [macro-stdexten] exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Voicemail2(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce exten => s,3,Goto(default,s,1) ; If they press #, return to start exten =>

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 MOH up to now have been using MP3 files and going

Hi Folks, I'm trying to interface * with a PBX, but seems that his ring cadence is somewhat different, and my T100 doesn't show any call coming in. I've tried to change zaptel to new values but still couldn't make it work. Is there any other place where I should be changing some parameter? Is there any tool to measure the cadence timing that this pbx is providing? Thanks!

Hello, I am not entirely sure the subject line captures what I am trying to do, but hopefully this description of the problem will help folks to see my challenge and hopefully offer constructive assistance. I have an experimental setup where I measure the decrease in oxygen in small vials as an organism, such as an oyster, consumes the oxygen. Each vial is calibrated before the experiment and

OK, just went through the research on how to set up scalable LDAP backends. By scaleable I mean without having Samba use the root dn to access ldap. This way, if you are going through the logs, you will be able to tell which domain controller is doing what. As far as I know, it *HAS* to be done this way because the posixGroup schema is way out of date (it wont take a dn as a member). This

Hey all...I'm brand new to * and I want to convert my home into a pbx type setup. I've figured out that I want a Wildcard X100P to bring my single POTS CO into my Linux box. My problem is that I'm sure sure what I need to do to get my analog phones connected up into a structured phone system. It *looks* like I can go the route of the Cisco Analog -> VOIP for about $100 on ebay.

Hello all, I've got a T100P connected to an Adtran TA750 with a T1 crossover... This connects to a patch panel with phone ports. The Adtran is fully populated with FXS cards. All I get on any phone port is a fast clicking noise... No dialtone. Asterisk 'sees' the card, (the channels show up in /proc/zaptel). Incoming calls are routed to the zap/x channel, but no ring. I'm

Hi, in the Snom FAQ I found the following information: After staring up, the phone tries the URL given in the "Setting URL" of the phone. ... BTW this setting can also be set via DHCP. .... option tftp-server-name "http://192.168.0.9/snom200{mac}.htm" The documents used: FAQ-04-06-14-sf.pdf "Setting up DHCP for snom phones" FAQ-04-03-24-sf.pdf "How can I

Hello all- So I have * up and running and connected to a legacy system via em_w lines and have no trouble dialing from * through the tie line but from the PBX across the tie line I am having intermittant receipt of the DTMF. T-Berd testing is showing that the digits are coming across but * is either missing the first digit consistantly. This seems to me to have something to do with start timers

Is there any way to receive and send faxes using a T100 card? If so how is it done? Gene Kochanowsky Solution Sciences, Inc.

Hello. I have two snom200 ip phones and 1 mp108fxs (audiocodes 8 fxs) and i dont get a ring in the caller phone when I dial from a snom200 to the other snom200 or the mp108fxs, I made a debug with ethereal, and I can see a "Ringing" packet being return from the called snom200 or mp108fxs to the asterisk box, but it is not being re-transmitted to the caller snom200. Altough

Hi, I have a data file which is named test.txt as below. Prior to that, I have converted the last row from nominal to numeric using as.integer. Statement T1001 T1002 T1003 T1004 T1005 T100 T1014 T1021 T1022 T1023 1 0 0 0 0 0 0 0 0 1 0 0 2 0 0 0 0 0 0 0 0 1 0 0 3 0 1 0 0 1 0

I am using the snom 200 with Phone type snom200-SIP Version snom200-SIP 2.04g Bootloader URL http://www.snom.com/download/snom200-boot1.9.bin Firmware URL http://www.snom.com/download/share/snom200-2.04o-SIP.bin I am using asterisk stable tree. I had to disable "Challenge Response on Phone" on my snom; I could not get it to work with

Hello, Does anyone know how to conference a call on the SNOM 200 phone? Whenever I push the cnf/tran button it just hangs up on the active call. The manual says you have to push the cnf function key but it doesn't appear in the lcd on my phone. Thanks -Matt -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have more or less decided to do with the SNOM phones for the next stage of testing with Asterisk becasue they seem to be the best value for money and have support for the GSM codec and easy upgrades.. But now I have to decied wheather to get the 100 or 200 and if its the 200 then I need to have some justification for the extra cost.. Can someone who knows these 2 phones tell me what the

Check the Redirection on the web interface of your Snom 200. If it says "When Busy" that's your problem. It should say "Never". Also make sure on Sip->Lines your line appearance says "All" or else you will have the same problem. Hope this helps :) Greg Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/3efdeb9f/attachment.htm -------------- next part -------------- Hello Though my SNOM 200 phone receive a voice mail but it doesn't show MWI on the LCD panel, Instead it keeps displaying "DND SW-REG,Call-log". Though I can access my voicemail using exten

Hello list, i am using: asterisk CVS-10/13/03-11:54:33 chan_capi-0.3.0 ATA-186 V2.16.1.ms over MGCP Situation: ISDN calls ATA ISDN speaks with ATA ATA-Phone presses Flash and speaks to another one (SIP/snom200) ATA-Phone hangs up ISDN talks to SIP/snom200 snom200 hangs up The incoming extension of ATA keeps busy for a time (20 sec?), even its not off-hook anymore! Any ideas? -- Swapping

Hi there, I tried to get a few "Optipoint 400 SIP" working with *, but it refused to work properly. In my testing-network i have two Sjphones (they are working really fine) and three optipoints. I?m able to dial the number of a Sjphone on all of the optipoints. The call is signalled at the Sjphone with the right number of the optipoint and an connection can be established. But when I