similar to: SVN update

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

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 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 =>

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

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

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

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 would like to setup a test number, that speaks back my phone number. How can I set this up? 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

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

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

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

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

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

The person on 617 is unavailable --- Why???? *CLI> -- SIP Seeding peers from Astdb: '617' at 617@192.168.250.107:6990 for 3600 -- Executing Dial("SIP/601-f18a", "SIP/617|60|tr") in new stack -- Called 617 -- Got SIP response 480 "Temporarily Unavailable" back from 192.168.250.107 -- SIP/617-602e is circuit-busy *CLI> sip show

iax2 show registry: Host Username Percived Refresh State ... 65.39.205.121:4560 511208 <unregistered> 60 Request Sent why it shows unregistered? I cannot make any calls to FWD anymore, ... Any idea? bye Ronald

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]

Ronald Wiplinger wrote: > Steve Totaro wrote: >> Ronald Wiplinger wrote: >>> kevin ling wrote: >>>> Hi, >>>> >>>> Check the vm_general.inc file >>>> >>>> >>> Where should this file be? >>> >>> >>> bye >>> >>> Ronald Wiplinger >>> >>> >> You

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

Can anybody tell me how I can set-up multiple access points with overlapping coverage, so that a moving WiFi phone user can continuesly use the phone. bye Ronald Wiplinger