similar to: call file mechanism

Hello, I have asterisk 1.2.9 running on a Debian sarge server, my outbound dial plan looks something like this: [outbound-longdistance] exten => _91NXXNXXXXXX,1,Dial(${OUTBOUND1}/${EXTEN:1}) About every other outbound call we make, we get the 'you must first dial a 1' message from our phone provider. It only seems to happen every other try if we try to make multiple out bound

Greetings all, I am in need of professional insight. I have a 2 node cluster running CentOS, mysql, apache, etc. I have on each system a fiber HBA connected to a fiber SAN. Each system shows the devices sdb and sdc for each of the connections on the HBA. I have sdc1 mounted on both machines as /quorum. When I right to the /quorum from one of the nodes, the file doesn't show up on the

Hello folks. I'm new to the list and I have questions about Samba. I have been able to configure Samba 3.x on Solaris 9 with AD authentication for the users. I'm able to mount the shares onto Windows XP clients and able to read the files. Now, if I use a text editor like notepad or GVIM to save an existing file, it saves it. When I try to use Word, Eclipse, or Crimson Editor, it

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

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

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'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

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, 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 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

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

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

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

I have been pondering the following... Voipbuster used to announce the cost of the call, but the new SIP servers do NOT. Because there's the choice between free (VoipBuster) and non-free (ADSL), I'd like to let the user know which one is actually being used by announcing it before the actual call gets connected, ie immediately after the channel proceeds from setup to actual