similar to: I am looking for some features,

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) exten => ronald,2,Voicemail,u615@office exten =>

Below is my extensions.conf for the fax part [incoming_28345474] ; ;******************************************************************** ; BEGIN - Inbound call handlers ;******************************************************************** ; exten => 8862100,1,NoOp(${CALLERID(num)}) exten => 8862100,2,Background(if-u-know-ext-dial) exten =>

I try to install astcc. Make install gives me: DBI connect('database=astcc;host=localhost','astcc',...) failed: Unknown database 'astcc' at ./astcc-admin.cgi line 67 DBI connect('database=;host=','',...) failed: Access denied for user: 'root@localhost' (Using password: NO) at ./astcc-admin.cgi line 60 Ignoring that (!!! ;-) ) and going to the

I tried to add the databases from the localhost to the database server and changed the every /etc/asterisk/*.conf from host=localhost to host=192.168.10.10 (my dababase server) When I restart asterisk, I do not get any errors, but after a phone call I see: Jun 7 18:11:56 ERROR[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

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

My database machine is broken and I have to use another one. I made somewhere mistake(s) and get now in the debug file: [Feb 24 09:05:24] DEBUG[32664]: MySQL RealTime: Query: SELECT * FROM sip_buddies WHERE name = '886' [Feb 24 09:05:24] DEBUG[32664]: MySQL RealTime: Query Failed because: Can't find file: './astconf/sip_buddies.frm' (errno: 13) [Feb 24 09:05:25]

vpbx*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 621/621 192.168.250.76 D N 5060 OK (65 ms) 626/626 192.168.250.109 D N 5060 OK (180 ms) 616/Ronald Softphone (Unspecified) D N 0 UNKNOWN 615/Ronald office 192.168.250.103 D N 5060 OK (41

I am looking for a way of not to install a softphone, preferable as a link on a web site to a webphone on MY SITE !!! Has anybody an idea for that? AJAX? bye Ronald Wiplinger

Skype is very succesfsfull and get more and more users, ... we can ignore them, accept them or 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

What does it mean? Jun 19 20:18:39 NOTICE[17484]: chan_sip.c:5472 check_auth: stale nonce received from '<sip:8862270@vpbx.elmit.com>' bye Ronald

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

before I accuse somebody to "overbill" I would like you to calculate with me: Rate: 0.0189 for calling Taiwan via NuFone Duration: 930 seconds Lets vote for the answers: 0.7269 or 0.2929 ??? bye Ronald Wiplinger

I got two phones on an ATA-186 (601, 602) and two phones on the TDM22B (603, 604). I have two lines on the TDM22B. I cannot figure out some of the problems: 1. 601 dials via ZAP/3-1 to local phone number at PSTN: ringing pickup on PSTN (empty) still ringing in the phone set 601 2. call from PSTN back: 601 picks up ... everything works !!! No caller id shows up 3. For testing I have only one

> 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-ascii; format=flowed > > I am looking for a

I have now a very strange situation. Asterisk dies exactly every hour at hour:09 !!! crontab has the entry: 10 3 * * * root /usr/sbin/asterisk-restart >/dev/null 2>&1 It seems that there is a time difference between mail server and asterisk server so that it might be "synchronized" to the crontab entry. *CLI> show version Asterisk CVS-HEAD built by root@vpbx on a

Calling from PSTN let extension 601 ring twice, hang up and starts over again to ring twice, ... If I pickup I do not hear on extension 601, and on the PSTN it is still signaling to ring. Can anybody enlighten me, please? extension.conf [incoming_88097074] exten => s,1,Wait(1) ;wait to get caller ID in. exten => s,2,Dial(SIP/102,20) exten => s,3,Voicemail(u102) exten =>

How should be the macro rewritten? [macro-faxreceive] exten => s,1,Set(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif) exten => s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN}) exten => s,3,rxfax(${FAXFILE}) exten => s,103,Set(EMAILADDR=ronald@elmit.com) exten => s,104,Goto(3) ... [Jan 23 10:43:38] -- Executing Macro("Zap/3-1", "faxreceive") in new

> -----Original Message----- > From: Ronald Wiplinger [mailto:ronald@elmit.com] > Sometimes it is not the "if" you make a search, often is for > new comers > "what" to aks for. > If you do not know the specific term, than you need to ask somewhere, > and I think the list is good for that. Sure. So say, "I tried a Googling for X, but I didn't

I try to change something in ASTCC, but I am now totally blind, .... I hang on one line now. I changed: vpbx:/var/lib/asterisk/agi-bin # diff astcc-original.agi astcc.agi 22c22 < # exten => _00XXXXXXXXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN}) --- > # exten => _00XXXXXXXXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${TARIFF},${EXTEN}) 35c35 < # exten =>

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