search for: phoneboy

...hough I've tried putting allow=ulaw first and it still didn't help) grover*CLI> iax2 show channels Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format 69.73.19.178 (None) 00003/00000 00001/00000 00000ms 0000ms 0000ms UNKN 69.73.19.178 phoneboy 00005/00102 00019/00017 00099ms 0000ms 0010ms UNKN 2 active IAX channel(s) grover*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 10.0.0.250 53 2b890d18-49 00101/00103 ILBC 1 active SIP channel(s) Is there a problem with iaxtel? An...

I am considering using Sipura-3000s as FXO devices for my * system. Has anyone tried them in that configuration? They interest me because they need no PCI slots and therefore no drivers. I would much prefer not to have any special kernel requirements for my system. /carmi

Since the 3000 has been out for a little while, has anyone done a review of the product? (couldn't find anything on google for wiki). Can the fxo and fxs ports be used as two independent channels? Is it really read for prime time? Etc. Rich

I don't see anything on the Wiki or in the documentation about disabling this feature.

I've been using * CVS code from May of this year and was able to connect to iConnectHere and receive calls with * being behind NAT. Now that I've upgraded to 1.0 RC1, this no longer works. I've tried setting nat=yes in places, externip, et al with no success .. even though the code I was running from back then worked without that. Any suggestions? BTW, I've gotten DTMF from

Anyone else got a Sipura 3000 in the UK? Apart from CID not working it also seems to not notice any of the line state changes on the PSTN when the remote party terminates the call. It only recognises the offhook signal which gets sent much later. Chris

My SPA-3000 finally arrived and I'm trying to get the FXO port on it to work as if it was a X100P card as far as Asterisk is concerned. I have Asterisk dialing out over the SPA-3000 FXO port no problem. The issue I'm having problems with is having the SPA-3000 automatically forward all incoming PSTN calls to the Asterisk "mainmenu" context (or ext I guess). Currently the

Hi List, I hope this setup must be done by our astersik users.. I am using Sipura 3000 to receive PSTN calls and forward those calls to asterisk for voice processing and after that, I am transferring call to extension through FXS port on SPA 3000. Currently, media of call is trombone through asterisk. i.e achieving blind transfers on asterisk with SPA 3000. Is it possible to stop trombone

I'd like to announce the time when people call and hit my voice-menu context, but the complaint is that the time announcement isn't interruptable. Is there any way to make SayUnixTime interruptable? -- PhoneBoy

...e through asterisk. i.e achieving blind > transfers on asterisk with SPA 3000. > > Is it possible to stop trombone and send back media path on Sipura FXO to > FXS. Basically you want to eliminate the back and forth traffic between the Asterisk server, right? That'd be nice. :) -- PhoneBoy _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

I am trying to configure the FXO port on a Sipura-3000 for use with Asterisk. When I connect to the Sipura to dial out on the PSTN line connected to the Sipura's FXO port, it gives me the dialtone of the PSTN line and then I can hear the DTMF for the number I dialled beforehand. It does work but the customer perceives this delayed second DTMF feedback as "unprofessional" and the

Just received: Cognigen is very proud to announce the official launch of Broadvox Direct, a new VOIP service. Broadvox Direct offers unlimited calling plans to anywhere in the US and Canada for a low monthly payment starting at $29.95 and basic accounts as low as $12.95. http://cognigen.net/broadvox/?mu Anyone know who's behind that? It's not BroadVoice, is it?

Sat Mar 22 14:16:55 CST 2003 This post is a bit long, but I want to make sure I am providing the information up front that can help in others helping me solve this mystery. I am having a bit of difficulty getting Shorewall to work with SecuRemote and its FW-1 server. I have attached the "rules" file I am using and the output of "shorewall show nat". The diagram below

The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk "..the most powerful tool for manipulating and accessing CPN data.." > http://www.theregister.co.uk/2004/07/07/hackers_gut_voip/ I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time.

I created an account at IAXTEL.com to route 1-700-XXX-XXXX calls through. IAXTEL.com gave me a number (example) of 700-555-6226. I have made the following changes to my: /etc/asterisk/extensions.conf: [iaxtel700] exten => _81700XXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1}) exten => _81800NXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1})

Hello, I have a problem with connecting a Digium X100P card to a Brazilian analog line. Can somebody help me out with this problem? My /etc/zaptel.conf is loadzone=br defaultzone=br fxsks=1 My /etc/asterisk/indications.conf [general] country=br [br] description = Brazil ringcadance = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 congestion =

I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It installed pretty easily and has worked great so I went to order some more of these units today. When I logged into Tech Data this morning, the PAP2-NA was now marked as discontinued and no longer available and only the PAP2 version was available which is the Vonage branded version. :( I saw someone on the list say that