search for: elmit

I have added in my zone file; _sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com. As I understand it should mean that any sip connection to <anyname>@elmit.com should go to the udp port 5060 at the host vpb.elmit.com. In Asterisk's extensions.conf I have in the context [default] exten => ronald,1,Dial(${PHONE_615},60,tr) ex...

...Database creation failed! Looking into the apache error log file I find: [Sun Feb 27 10:45:50 2005] [error] [client 192.168.250.108] DBD::mysql::st execute failed: Table 'astcc.brands' doesn't exist at /srv/www/cgi-bin/astcc-admin/astcc-admin.cgi line 564., referer: http://vpbx.elmit.com/cgi-bin/astcc-admin/astcc-admin.cgi?mode=Home [Sun Feb 27 10:45:51 2005] [error] [client 192.168.250.108] DBD::mysql::st execute failed: Table 'astcc.brands' doesn't exist at /srv/www/cgi-bin/astcc-admin/astcc-admin.cgi line 163., referer: http://vpbx.elmit.com/cgi-bin/astcc-admi...

I try to setup SJphone on my PDA, but it does not register with Asterisk. I have setup a sip account on asterisk, ... Can anybody give me a hint? bye Ronald

...[7877]: cdr_addon_mysql.c:400 my_load_module: Failed to connect to mysql database cdr on 192.168.10.10 Or if I try to register a phone, which is in the sipbuddies. Jun 7 18:14:43 NOTICE[7870]: chan_sip.c:8098 handle_request: Registration from '"Ronald Video (612)"<sip:612@vpbx.elmit.com>' failed for '192.168.10.18' What do I miss? bye Ronald

I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald

I tried to use message waiting indicator, by "Subscribe for MWI" in the web menu of the phone. However, it does not light up / flash, even if a voice mail is waiting. Where is the switch to turn it to? bye Ronald

> From: Ronald Wiplinger <ronald@elmit.com> > Subject: [Asterisk-Users] Is there hardware to remote control > available? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <41C6D43F.5070201@elmit.com> > Content-Type: text/plain; charset=us-as...

Earlier this afternoon I had this working exten => 2890,1,Answer exten => 2890,2,GoTo(12) exten => 2890,12,Wait(1) exten => 2890,13,Festival('I can say numbers like') exten => 2890,14,SayNumber(1230001,f) exten => 2890,15,Wait(1) exten => 2890,16,HangUp I was so very proud of myself... All of a sudden after a reboot.... I get the following from the same call plan

I tried to install astguiclient and it gives me for each follow page: Object not found! Looking into the apache log file I find: [Sun Feb 27 16:18:30 2005] [error] [client 192.168.250.108] File does not exist: /srv/www/htdocs/astguiclient/method=POST, referer: http://vpbx.elmit.com/astguiclient/phone_stats.php?extension=gs102&server_ip=192.168.250.20 I have uncommented the ### If you have globals turned off uncomment these lines $PHP_AUTH_USER=$_SERVER['PHP_AUTH_USER']; $PHP_AUTH_PW=$_SERVER['PHP_AUTH_PW']; $ADD=$_GET["ADD"]; ### AST GUI dat...

I would like to know which extension number is engaged in a call. show channels shows me: *CLI> show channels Channel Location State Application(Data) SIP/asterisk.elmit.com-0 690@default:2 Up Echo() SIP/8807-066 690@newcontext Up Echo() 2 active channels 2 active calls but it is not true!!! show channels verbose gives me even a time to each of 112:43:33 and 347:23:22 I want to know: 1. is extens...

I upgraded recently to Asterisk SVN-branch-1.2-r25165M the commandline asterisk -rx 'sip show peers' returns with the first line: on Is that a bug, or how can I omit it? I used: asterisk -rx 'sip show peers'|grep OK|sort | tee /dev/tty |wc -l; echo " registered at ELMIT" which results (because of on) to: Binary file (standard input) matches 1 registered at ELMIT

...do something,... My suggestion is that we try to do something, ... If we would peer to each other, than we get soon also a great amount of users together, and than our service becomes more valuable, ... Let's discuss advantages and disadvantages! bye Ronald -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com +886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message ba...

I have a very simple question . I have 2 internal extension 301 and 300 sip phone . I want to these extesion can call each other, and ext 300 can call outside to pstn, and ext 301 to call internatonal. How can I do that ? [x1] exten => 300,1,Dial(SIP/300) include => pstnlocal [x2] exten => 301,1,Dial(SIP/301) include =>international [pstnlocal] exten =>

Is there a way to test if the motherboard is ready for a Digium card (PCI 2.2) ? I would like to know from a remote computer, where I have (root) access, if this computer is ready for a TDM22B. bye Ronald

...s,2,Set(EXTEMAIL=${DB(${MACRO_EXTEN}/xEmail)}) exten => s,3,NoOP() exten => s,4,Set(EXTNAME=${DB(${MACRO_EXTEN}/xName)}) exten => s,5,NoOP() exten => s,6,Set(EXTCOMPANY=${DB(${MACRO_EXTEN}/xCompany)}) exten => s,7,rxfax(${FAXFILE}.tif) exten => s,103,SetVar(EXTMAIL=ronald at elmit.com) exten => s,104,Goto(7) exten => s,105,SetVar(EXTNAME=Ronald) exten => s,106,Goto(7) exten => s,107,SetVar(EXTCOMPANY=Elmit) exten => s,108,Goto(7) When I call this PSTN number and dial the extension number 3000, then I see that: *CLI> [Jan 27 16:03:21] -- Zap/3-1...

Hi all, a noob here, I am trying to get outbound calls through asterisk working with Broadvoice. I have consulted the following two online tutorials: http://www.broadvoice.com/support_install_asterisk.html http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice in an effort to make outbound calls. My current settings are as follows: sip.conf register =>

I bought three plus two Grandstream BudgeTone 101 phones. The shipping cost more than the phone itself from Pulver store. The first shipping had one phone defect. Nothing on the display. (Can happen!) The second shipment had one phone with a defect display, but it still worked. The second phone's handset was defect too (microphone did not work). Changing the handset from this one to the

I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger

Hello All, Yep I'm a newbe. I'm just started to play with asterisk. What I have Redhat Fedora Core 2 (New install) 3 X100P cards. I installed zaptel-1.0.3 libpri-1.0.3 asterisk-1.0.3 Where should I start?? -- Thanks, David -- This message has been scanned for viruses and dangerous content by KE6UPI, and is believed to be clean. KE6UPI thanks MailScanner for their support. Please

hi, i am using Asterisk CVS-05/31/04. i have the problem that sip clients can make calls over asterisk without registering befor. the xlite is not loged in with any username/secret bit still can make calls over asterisk. how can that be? thx for help. thomas