similar to: Forwarding SIP over IAX problem: No One Available

Hi, Is it possible to use asterisk with Free World Dialup (FWD) ? Did someone manage to make it work? how? Best, -P -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup CareerBuilder.com has over 400,000 jobs. Be smarter about your job search http://corp.mail.com/careers

Hi, I can't get dial-out working. I'm trying to call 523936. Is there something wrong with my setup here? Could someone please give me a few pointers? Regards, Norman Zhang [fwd-out] exten => _8.,1,SetCallerID(${FWDUSERID}) exten => _8.,2,SetCIDName(${FWDUSERNAME}) exten => _8.,3,Dial(SIP/${EXTEN}@fwd,70) exten => _8.,4,Macro(fastbusy) exten => _8.,5,Hangup *CLI>

Hi, Has anyone managed to create a setup with a Sipura SPA-1001 as a client? Right now I can connect to the device by dialing the extension number but when I try to connect from the phone handset to make an outbound call it gives an unavailable tone. I'm using Line 2 on the SPA-1001 to connect to the local asterisk server, line 1 is used to connect to my VOIP provider until I can get the

I have a TDM4xx card with two (3 and 4) interfaces for my land lines. I have a basic setup working with them and one VoIP provider. Questions: 1. How do I determine which Zap line the incoming call is on so I can handle it differently? One line is my home phone and the other is my work line. I would like different dialplans for each. 2. When I have my work line set (via Verizon) to call

who can tell me how to do TDM over enthernet ? pc a connect pc b only use TDM card? thank you John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?23? 11:47 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5, Issue 336 Send Asterisk-Users mailing list

Hi all, Here is a graphical diagram of what I am trying to do: <SIP> <---> <GW/NAT/*> <--IAX--> <*> <--TDM400P--> <Analog Phone> So I have incoming SIP calls go to the * on the GW, which I then want to forward over IAX to the second * box behind the NAT GW. If I was to place a call on the second * box, it should then forward to the * on the NAT GW

Hello. I have another issue. When I call in, everything is processed correctly, including voicemail, but I don't hear any ringing/ringback. exten => s,1,Zapateller(answer|nocallerid) exten => s,2,NoOp exten => s,3,Playback(pls-wait-connect-call) exten => s,4,Dial(${PHONE1}&${PHONE2}&${PHONE3}&${PHONE4},15,Ttm) exten => s,5,Answer exten => s,6,Wait(1) exten

I have installed my TDM11B according to the docs at the Digium page but I do not get incoming caller id. My telco confirmed that callerid should be passed but I do not see it coming in. I am in The Netherlands with a KPN line. The number is not even visible in console mode on * running stable 1.0.5. Ideas anyone? -- Starting simple switch on 'Zap/4-1' Feb 1 19:19:40 NOTICE[16582]:

Hi, I wish to connect several ATA186 Phones to each other, to iconnecthere and to the PSTN using asterisk. Please tell the appropriate settings for firewall (ports to open etc.) sip.conf and extensions.conf(part relevant to iconnect). Also I would be glad to get a working example of your ATA186 configuration. I tried searching the mailing lists and several sites but did not find an answer.

Hi, I had a FWD account set up with asterisk (using SIP) and it was working fine both ways. I switched to IAX2 and now I can't get incoming calls from FWD. People who call my FWD number get a "480 - user is not online" message without any traffic reaching my box. I can call FWD numbers fine over IAX2. It seems fwd isn't trying to place the call over IAX2 because it thinks

Hi, I have an X100P which receives an analog line from another PBX. These are the relevant entries in extensions, PHONE1=Zap/1 [macro-extensions] exten => s,1,Dial(${ARG1},20) exten => s,2,Voicemail2(u${ARG2}) exten => s,102,Voicemail2(b${ARG2}) exten => s,103,Hangup [home] include=>tozap exten => 2201,1,Macro(extensions,${PHONE1},${PHONE1VM}) exten =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a medium sized business that is interested in implementing * as their PBX system. They currently have a Panasonic system with Panasonic handsets that they are going to replace Asterisk with, as the current system is maxed out, and they don't even have voicemail capabilities. I have been considering using an Adtran Atlas 550 with FXO and

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 OK, so I have a local extension that a phone can call to take it to voicemail. I don't want it to exit out to a fast busy tone, as I would rather it allow the user to simply continue on and call a new number (without having to physically release the line first). The [intern] context is where everything goes by default (sip.conf for example has

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I am using a TDM40B, and have managed to compile mpg123 and turned on MOH. Problem I am having is that it is choppy, staticy, and sounds like white noise pretty much. I have search the archives to see if this problem had been resolved, but I haven't found anything yet. Has anyone had this problem and resolved it? I am calling from

Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?24? 7:51 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5,

Hi All, I have just compiled the newest version of mpg123 on a RedHat 9.0 system (mpg321 has not been installed) and I am using the newest CVS version of asterisk. Whenever I place any mp3 files in the /var/lib/asterisk/mohmp3/ directory, Asterisk crashes a horrible fiery death. If mp3s exist in that directory, then I can't even start Asterisk. If I start it without files then copy

Hi, I got my * working fine with FWD at office with 2 extensions, i receive calls and i can make calls thru FWD. I got also my * at home, and i connected it using auth=rsa. From my home, i can make calls using my office iax, but if i try to redirect incomming calls from FWD to my * at home, it rejects the call. I created the pub/key pairs for rsa and its working ok and i just pasted the

I'm just curious if I was to place my * box behind a a FW/NAT box running linux, if my SIP calls will still work. Box right now is a RH9 computer using iptables as the FW. I wouldn't mind placing my * box behind it, but I'm wondering if anyone has actually gotten NAT working with *? Thanks, -- +------------------------------------------+ |Leif Madsen -

I'm sorry this is somewhat offtopic, but I do plan to use this to help me create documentation for the * project.. so I guess it is somewhat on topic :) Anyways, I am looking for some sort of document control system. It should act somewhat like a CVS where it keeps previous versions, allows people to submit documentation, keeps track of who has what document open etc.. etc.. The

I can't get my Asterisk to register/place calls with FWD. Here's what I have in my SIP.CONF: register => 11111@fwd.pulver.com/11111 [fwd] type=friend secret=somesecret host=fwd.pulver.com username=11111 fromuser=11111 fromdomain=fwd.pulver.com I'm using CVS version of Asterisk, checked it out last week. I get authenticate error when registering with fwd, and all my calls to