similar to: Any other "free" toll free SIP providers out there?

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

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

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, I can't receive/send calls with Asterisk. Could someone please give me a few pointers on my configuration? Regards, Norman Zhang ; sip.conf [general] disallow=all allow=ulaw port=5060 bindaddr=0.0.0.0 externip=x.x.x.x localnet=192.168.22.0 mask=255.255.255.0 context=inbound-sip maxexpirey=180 defaultexpirey=160 tos=reliability srvlookup=yes register =>

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",

I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX

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

Sorry about this, I have been struggling with the basics of my asterisk config. I set up two sip peers and two phones. And I set up lots of dial masks for outgoing calls, all my outgoing calls were working great, however incoming calls were a different matter altogether, I cannot get incoming calls to work. So I have gone back to a very basic FWD config, with one phone which as far as I am aware

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,

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 =>

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

We all know the FWD is NO more available. How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite ? Any alternative for FWD ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090820/4206395a/attachment.htm

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

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

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

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

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

Hi, I have some doubts on sip.conf. 1) Can I have two or more SIP phones acting as extensions in one Asterisk box, and at the same time, registered to a SIP proxy, say Free World Dialup? If yes, how? 2) Why we need a section in the sip.conf for the proxy, say, Free World Dialup's fwd.pulver.com? In the case of 1), how to assign the value to section [fwd.pulver.com], since there are more

Hi :) I've had all this working before, but I'm revisiting it, and in short, I currently have huge problems receiving incoming calls. I've been trying with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel and libpri as of yesterday afternoon. Would someone mind helping? :) My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set as the 'DMZ