similar to: Voipbuster not ringing, other SIP peers are ringing...

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

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

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

Hello asterisk-users, Just curious if anyone has the indications for Ireland, tried googling for it to no avail. Sean -- +---------------------------------------------------+ |VOIP= FreeWorldDial: 689482 VOIPBUSTER: thecivvie | |GPG Key http://thecivvie.fastmail.fm/thecivvie.asc | +---------------------------------------------------+ Strange things happen under the midnight sun when Men

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

Hello, I want to know if is posible to call from a mobile to my Asterisk (national call) and then I insert an international number in my mobile and Asterisk call with a Voipbuster account. Is this possible? Thanks a lot. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070725/206f52be/attachment.htm

What do I need to install for OSS or ALSA for console use only Sean -- +----------------------------------------------------+ |VOIP: FreeWorldDial 689482 VOIPBuster thecivvie | |GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc | +----------------------------------------------------+ -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type:

Hello, I'm trying to receive a fax to my inbound number from voipbuster. Asterisk receives the call and starts the rxfax application successful, but then nothing happens. The calling party is still hearing a ringing tone, or sometimes nothing. Voicecalls are working correct and without problems. For testing I've add a local number (300) to the dialplan. When I call this number

Hi, all I am reallty pissed with their service. I wonder if this is common problem. Firstly, all of my calls are terminated after 30s. And termination happens in a strange way. My local asterisk server does not see the disconnection, but remote party is disconnected. Basically, I am still on the phone, while remote party was disconnected. When I hang up, I get something like that: Apr 20

I've got a strange situation with musiconhold. It works if I dial my extension 6000: >From extensions.conf: exten => 6000,1,Answer exten => 6000,2,MusicOnHold() Debug output if I call 6000: -- Executing Answer("SIP/gs1-b6ee", "") in new stack -- Executing MusicOnHold("SIP/gs1-b6ee", "") in new stack -- Started music on hold,

Hi;? When connecting via VoipBuster or VoipStunt, "I can hear them but they can't hear me"?. This happens with VoipBuster or Voipstunt. Registration is done correctly. I thought it could be something related to NAT, but I don't have this problem when using VoipUser or Asterisk2PSTN, for example. ? I tried with different codecs: gsm, alaw and ulaw but no change. ? So, now?I

I would like to put srvlookup=no in my SIP conf, so that I don't get DNS issues (Asterisk stops responding). I use VoIP Buster and in sip.conf I use sip1.voipbuster.com. When I do sip show peers in CLI I get voipbuster/tomo 194.221.62.207 5060 OK (27 ms) And when I ping sip1.voipbuster.com [root@tomo ~]# ping sip1.voipbuster.com PING sip1.voipbuster.com