similar to: Name/username (sip show peers)

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

How to set-up LCR ? a. which companies can be used with LCR? b. how to set-up & maintain LCR? c. multiple connection to one gateway? Example: +886223456789 could be reachable via a. ENUM free b. Dundi free c. Voipstunt free d. Voipbuster free e. Nufone $ f. Voipstunt $ g. others with 4 concurrent connections $$ h. others with 3 concurrent connections $$ I am looking

Does anybody else see the same behavior for VoipBuster connections? When I trace one of the other SIP peers, I see it sends this message: ---------------------------------------------------------------------- <--- SIP read from 82.101.62.99:5060 ---> SIP/2.0 180 Ringing Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE Call-ID: 740540ee64fa957513ce89f03ef5e3f2 at sip.xs4all.nl

I have something like below in my sip.conf. How can I put this into Real-time? [voipbuster] type=friend ; (or "peer" if we don't need incoming calls, or if there is a separate section with "type=user") host=sip1.voipbuster.com disallow=all allow=ulaw allow=alaw allow=gsm allow=g726 username=abcd1 ;={{YOURUSERNAME}} fromuser=abcd1

I got four phones: 601 is a SIP phone (no brand) 615 is Snom 190 621 is a Grand stream 628 is a remote SIP phone (no brand) 601, 615, 628 can call each other without any problems 621 used to be able to call remote 628, but after upgrade to CVS Head Nov. 11 the remote party cannot hear me. 615 never could call remote 628, both party hear nothing. 601 can always call 628 [Oct 16 00:52:13] --

Hi, I am new here on this list, and have a problem of which I hope that somebody here can help me with it. I have a Voipbuster account, with which I would like to make phone calls via my Asterisk PBX. If I let X-Lite register directly at voipbuster.com, everything is OK, but if I let Asterisk register there, it says "registration for XXXXXX@sip.voipbuster.com timed out, trying again",

Hi, all I am still battling to connect * and voipbuster. What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or IAX traffic when using their client. VoipBuster client connects to connectionserver.voipbuster.com on port 11112 for authentication. Call itself is placed on different server. I have tried to connect using SIP and IAX and it seems that no authentication is

Hi, I was successfully using VoIP Buster via IAX2 for several weeks now. Yesterday/today it spontaneously stopped working. Using the "real" client the connection works well though. Anybody else experiencing this problem? Or asked differently: Is there anybody for whom it is still working? Can anybody tell me what the problem could be from this: -- Executing

Hi, I'm using voipbuster at work, and I've got 2 questions: 1) Is it possible to send faxes using voipbuster connex? 2) Is it possible to cut off or cover the voice that say the charge per minute?(I've payed the '5' euro, and from that moment I've got it!). Of course I understand that is to let me know how much I'm going to spend, but I do not like it, expecially when

Hi, all Here is a something I found on the web: http://www.voipbuster.com And it works OK too. Now, I want to use it via asterisk, so I ccan use my normal phones instead of PC application. Did anyone try to connect astersisk and VoipBuster? Thanks, Rudolf

Hi, all, I got my * to work with voipbuster service. And it works quite well when I am calling USA or Europe. However, for local calls, I am experiencing long delays (About 1s). As far as I know, voipbuster application does not have this problem. I am using IAX and gsm codec. Any ideas on how to combat this? Thanks, Rudolf

I am trying to move from IAX2 to SIP for voipbuster, moving at the same time to sip1.voipbuster.com. When I try calling out, I see that there is SIP exchange, and in many cases also RTP data being exchanged. Hover in a very large number of attempts the connection is not established. Half of the time there is no RTP, the rest of the time there *is* RTP data flowing in two ways, but no ringtone is

Hi, I have the problem that WaitExten is not responding to key presses. Here are the sections from my extensions.conf: [globals] incoming_call=0 menu=0 announce=0 [internal] exten => 777,1,Goto(hotline,${EXTEN},1) [hotline] exten => _X.,1,Set(CALLERID(name)=Hotline) exten => _X.,n,Set(original_extension=${EXTEN}) exten => _X.,n,GotoIf($[${announce}=1]?4:10) exten =>

Hi All, This is more of a hardware question that an Asterisk question so I hope this is still the correct place for the post. I know with the Linksys phones you can create a hotline by using the dial string of (S0<:number>). I have been trying to do this with a PolyCom phone but I have not been very successful. Does anyone know how to create a hotline phone with a PolyCom?

*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

I have two phones (111 and 112) on a LAN, and I have on a users site a phone 333. phone 111 uses sip.conf, while 112 uses real-time set-up. 111 can call 333 AND the audio is working 112 can call 333 but audio is just white noise. 333 can call 111 or 112 and audio is working. The phones are identically set-up (just user name = phone number and password are different) sip.conf (for 111 - all

hello.... I'm new to wine and tested and installed it, because I wanted to run the hotline server from windows under linux... upload, news works great but download from this maschine is not working! strange thing does anybody has the hotline server runing? do you have had similar problems with another network app? this are the errors.... imted from C:\windows\system\wininet.dll, setting

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

Is there anyway to prevent an extension from dialing certain codes. ie I want to prevent extension 203 from dialing number which start with 00 087 086 etc Sean -- +----------------------------------------------------+ |VOIP: FreeWorldDial 689482 VOIPBuster thecivvie | |GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc | +----------------------------------------------------+ --------------

What settings are people using? I've seen the ones from dslreports but I'm in that lucky group of people that paid the 1 euro just to have it no longer work. Even after I setup a additional account over the weekend it still doesn't work. And, of course, etherreal only shows encrypted traffic so I can't snag any config settings from it. Any assistance? -----Original