search for: voipbuster

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 IA...

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", even though all settings are...

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

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: 740540ee...

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...

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 c...

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 di...

...om it. Any assistance? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tony Hoyle Sent: Monday, August 15, 2005 7:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? Erick Weber V. wrote: > For me to > Works for me... Tony _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options v...

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

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*...

...uot;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 Dial("IAX2/arik@arik/2", "IAX2/username@voipbuster/0049712147557") in new stack -- Called username@voipbuster/0049712147557 -- Call accepted by 213.61.187.156 (format alaw) -- Format for call is gsm -----takes a long while ~15 to 30 sec here------ -- Hungup 'IAX2/voipbuster/3' == No one is available to answer a...

Hi All, Any VoipBuster SIP users on this list that'd be willing to test VoipBuster outbound VoIP to PSTN? All numbers I tried from my (*) server are supposedly being connected, but no phone rings! Also their new WebStart function doesn't cause my phone to ring either... TIA! -- Francesco Peeters

...nd put a call on hold the other party hears the musiconhold: Debug output when I do an outgoing call: -- Executing SetCallerID("SIP/gs1-cb7a", ""Anonymous" <0031xxxxxxxxx>") in new stack -- Executing Dial("SIP/gs1-cb7a", "SIP/0031xxxxxxxxx@voipbuster") in new stack -- Called 0031xxxxxxxxx@voipbuster -- SIP/voipbuster-ac66 is making progress passing it to SIP/gs1-cb7a -- SIP/voipbuster-ac66 answered SIP/gs1-cb7a -- Attempting native bridge of SIP/gs1-cb7a and SIP/voipbuster-ac66 -- Started music on hold, class 'defau...

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 (194.221.62.206) 56(84) bytes of data. So, Asterisk is registered at 194.221.62.20...

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 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=ab...

Hi list, I have a call file as following and it works. But, I don't really understand its mechanism. The SIP/voipbuster is a sip trunk which I set up in freePBX with voipbuster account. And 2874 is one of my extension which was assigned to x-lite client. When I place this call file in outgoing folder, it is able to dial out my home phone at 001xxxxxxxxxx. However, the Dst in call logs show 2874 or s instead of m...

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 | +----------------------------------------------------+ -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 2859 bytes Desc: S/MIME Cryptographic...