search for: rumatech

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 I seem to have installed the G729 codec properly but can't seem to get it to work .. in the Asterisk startup I see .. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator

Hi, all I finally got myself one of those SIPURA boxes. It is labeled as Linksys, but this is actually a SP3000 box. Anyway, unit has lots of configuration parameters. Not all are obvious. At the moment it registers against my *, but all the calls I do from analog phone connected to it, go to VoIP channel. As this part is still in testing, I want all the outgoing calls got to PSTN by default

Hi, all Can someone tell me where to tell asterisk no to use 127.0.0.1 IP (localhost)? When I am registering with VoIP providers, they get my info as s@127.0.0.1. (This is SIP registration). Also, in SIP logs, when calling I am getting things like this: Executing SetCallerID("SIP/phone2-22c3", ""CID Name" <CIDNUMBER>") > in new stack > -- Executing

Hi, all I am building a new server. Have installed FC 6 and put in TDM400 card. Checked out latest asteriusk code, run make install in zaptel directory. So far all is fine. Now I am trying to install the drivers. # modprobe zaptel FATAL: Module zaptel not found. Fair enough, no zaptel driver is found on the system. Is there are any known problems with FC6? I did not have much trouble running

Hi, all I am trying to figure out how to forward incoming PSTN call on SPA3000 to VoIP extension(s). Basically, I have converted my home to VoIP. I have normal phone (connected to SPA3000) and couple of IP phones. All call coming from VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I need to do same thing for incoming PSTN calls. I have enabled gateway function in SPA3000 and

Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI> exit No such command 'exit' (type 'help' for help) *CLI> quit No such command 'quit' (type 'help' for help) *CLI> Any other ideas? I started asterisk with -cvvvvg option. Same problem if use asterisk -r to connect. Can not exit. Any

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

I am attempting to update my Asterisk installation from 1.0 to the latest stable version. When I use CVS checkout, I am receiving the following messages on chan_sip.c: RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v retrieving revision 1.510.2.25 retrieving revision 1.510.2.27 Merging differences between 1.510.2.25 and 1.510.2.27 into chan_sip.c M asterisk/channels/chan_sip.c Then, when

...> Subject: Re: [Asterisk-Users] Sipura SP3000 question To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <20060422191752.GR24768@roshan.info> Content-Type: text/plain; charset=us-ascii On Sat, Apr 22, 2006 at 11:19:35PM +1000, RumaTech scribbled: > As this part is still in testing, I want all the outgoing calls got to > PSTN by default and dial, say 0, to get an "outside VoIP line". > I would like to do it as part of SP3000 configuration, not as part of > * dialplan. Can someone help me? I use the following...

Hi, all A quick question. I have SPA3000 and trying to get call waiting to work. I do receive call waiting tone, however hook flash does not seem to work. I think, I set up SPA3000 correctly. Basically, doing HF 2 (switching calls in Australia) does not do anything. Is there any examples on how to setup hook flash operation in Asterisk? Thanks, Rudolf