similar to: How to start a playback after the called partypicks up?

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 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 =>

[public_gk] ;exten => _070.,1,Set(CALLERID(number)=070333333${CALLERIDNUM}) exten => _070.,1,Dial(H323/${EXTEN}@59.120.139.119) exten => _070.,n,Hangup *CLI> h323 show peers Name Accountcode ip:port Formats 7000 ast_h323 203.160.252.147:1720 0x4 (ulaw) 88670333333 ast_h323 203.160.252.147:1720 0x4 (ulaw)

I am still looking for a solution and I am sure that I am not the only one having that problem: If provider A fails for any reason, the next provider should be taken. There are many reasons, why a provider fails, like: password wrong (cli reports so, but actually it is the gateway's problem) gateway temporary not reachable gateway busy ... Our user places a call, the gateway responds with

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 =>

voipstunt: "Forbidden - wrong password on authentication for INVITE to ...." I have paid, the password was not changed, ... I have no idea why. Is there anything what I can do to get this "failed" call over to another provider, so that the user can complete the call? (Dialstatus was an idea, but the line does not show up in CLI) [Apr 5 09:22:36] -- Executing

Hello, To detect an answering machine I have found following two commands, BackgroundDetect (comes with asterisk) MachineDetect (asterisk add-ons) First question, does BackgroundDetect works well with g729? I havn't try MachineDetect yet, what is the benefit of MachineDetect over BackgroundDetect. If anybody used any of this command successfully, please help me. If possible, please let me

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

How can I make it more readable? Name/username 601/601 123456789/123456789 voipbuster/abcd 601 = hotline 123456789 = Peter Pan only voipbuster/abcd is easy read/understandable! 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

*CLI> show features Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # ## Attended Transfer *2 One Touch Monitor *1 Disconnect Call * *0 Dial option is tTwWr I tried to call from 601 to 615 601 keys in *0

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

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

Is anybody using XEN in conjunction with Asterisk and/or Openser? I would like to get some info about such an environment and experience reports. bye Ronald Wiplinger

I want to add some sound filed on demand during a phone call only possible on some extension numbers. I get many phone calls from local companies, but don't understand Chinese! I would like to record the call, but also ask the caller some questions, which should be added into the call with some keys on the phone, ... e.g. *66554 should add into the call: How are you? or What is your

If a user calls and hangs up before the destination party rings, than the in-use flag remains set! This is one case, but maybe there are many other cases. I have created a number the user can dial to reset this flag. However, that is written in the manual!!! Who reads a manual anyway!!!! I want to make to reset all in use flag with a program. Has anybody done it, or has a better idea? My idea

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

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

The time stamps in ASTCC are useless as they are now: Fri Jul 22 15:06:25 2005 Wouldn't it be better to use something like: 2005-07-22 15:06:24 Fri I want to sort the records by date, but with the format now it is impossible... or do I miss something? bye Ronald Wiplinger