similar to: Voicemail answers, but drops SIP call after about 3 seconds.

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

Hello I am trying to register for fwd from * but having problem and unable to solve it. I keep getting this message *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4800 (handle_response): Failed to authenticate on REGISTER to '<sip:89699@fwd.pulver.com>;tag=as62a7f29b' NOTICE[1125329600]: File chan_sip.c, Line 4800 (handle_response): Failed to authenticate on REGISTER to

Which phones do you have? We are using Cisco 7940G phones and I have been able to do this by modifying the dialplan.xml for the phone to rewrite numbers as they are dialed to include the "9" in front of whatever is dialed from the phone. Now you can use the received calls menus without having to edit the numbers before hand. Calvin On Jan 29, 2005, at 12:13 PM, Stefan Gofferje

Hello list, Can someone please point me out why would a stream like the following only write ONE line (the first) on the given file? Action: login Username: test Secret: 123456 Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-000000: Append Cat-000000: default Var-000000: 127 Value-000000: >5555, Jason Bourne97, jason25 at noCia.gov.do ActionID:

Hi Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten

I have a small HUGE problem with *. I have installed * but I have 2 problems. 1 - Making call to FWD. 2 - Receiving call from FWD More info of the problem at the end. Here is the sip.conf file. ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls register =>

My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP phone. Here is my sip.conf file: ; ; SIP Configuration ; [general] context=default ; Default context for incoming calls port=5060 ;added bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ;

I'm wondering if there's a way it will show on the phone when there's a new message. Here's what I'm trying to do : in my extensions.conf when someone call from a PSTN line on my TDM04B card they have a choice. When someone press 1 for sales then I have 3 phones ringing at the same time. Each phone as already there own mailbox because if someone know there extension

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to

In this dialplan, the SIP user agent is a Sipura two line adapter with line 1 as SIP ID "1000" and line 2 as SIP ID "2000". Basically I have this set up so that 1000 and 2000 are "lines in hunting" on incoming extension "555". I want an incoming call to try to ring ext. 1000, if 1000 is busy, then ring 2000, if 2000 is also busy than ring Voicemail. Here

Can multiple FWD accounts be registered? I have the following output in my sip.conf file: register=74928:xxx@fwd.pulver.com/74928 register=75160:xxx@fwd.pulver.com/75160 register=74573:xxx@fwd.pulver.com/74573 [fwd-74928] type=friend secret=xxx username=74928 host=fwd.pulver.com [fwd-75160] type=friend secret=xxx username=75160 host=fwd.pulver.com [fwd-74573] type=friend secret=xxx

Hey all, I'm trying to get my Asterisk server up and running on fwd.pulver.com just to get the hang of it until I get my FXO card in a couple of days. It seems to connect but that's about it. If I try to dial into it from another fwd # it says user is not online. In sip.conf I have the following added: register => xxxxxx:xxxxxx@fwd.pulver.com/489125 [fwd.pulver.com] type=friend

I'm hoping someone can help me out with this. I am basically just trying to separate my outbound and inbound calls into separate contexts instead of having everything in a single context. Any help would be appreciated. Perhaps I've missed something really obvious.... Here is the network layout: <remote> <--TDM400P--> <nattedbox> <--IAX--> <liveipbox>

Hi forum, I have been fighting days and days configuring FWD and asterisk with NO success I have the following scenario. My sister in Spain with FWD dialup client My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone. Spain LAN FWD

Hi All, Is there any way to connect (register, initiate and receive calls) with Asterisk to FWD through NAT? Since I own my router port forwarding is not a problem. I tried with Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwd.pulver.com but since Asterisk still use internal IP in some SIP fields I got "479 We don't accept private IP contacts. Please set your external

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

I tried it w/ mine as well and it hung up on me because I just have Voicemail running not Voicemail2. It seems as though you have Voicemail2 because it's trying to play the Unavialable message. Just a thought though. Does it do the samething w/ [qout-phillyq] exten => 0,1,Voicemail(u1) exten => 0,2,Goto(default,s,1) Tim Thompson http://www.amatechtel.com (806) 722-2227

Sirs, Thank you for your quick response. But when i try to make a call to FWD the following error appears: For example, when i call to 612 (a service number of FWD) -- Executing Dial("SIP/Phone4-e85b", "SIP/612@fwd.pulver.com|90|Ttr") in new stack -- Called 612@fwd.pulver.com -- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)"

I bought three plus two Grandstream BudgeTone 101 phones. The shipping cost more than the phone itself from Pulver store. The first shipping had one phone defect. Nothing on the display. (Can happen!) The second shipment had one phone with a defect display, but it still worked. The second phone's handset was defect too (microphone did not work). Changing the handset from this one to the

I've been having issues getting FWD to work. I posted this same Q to the FWD forum (no responses yet), but I was hoping someone here had some insight: http://yabb.pulver.com/cgi-bin/yabb/YaBB.cgi?board=news;action=display;num=1072263468;start=0#0 I just signed up for an FWD account (I know I had one before, but I lost it.. :) I've got it running through Asterisk - all working fine