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...ving 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 => 10082:mypass @fwd.pulver.com/201 [fwd.pulver.com] type=peer username=10082 host=fwd.pulver.com [200] ; This is the cisco ATA phone type=friend host=dynamic defaultip=192.168.0.167 dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info context=sip callerid="Yehiel" <10082> The same o...

...is handling the call as if it is going across the NAT ( see attached sip debug output). However (there is always a however) I get and the call fails. Oct 6 20:01:20 NOTICE[1087269568]: chan_sip.c:6766 handle_response: Failed to authenticate on INVITE to '"465605" <sip:465605@fwd.pulver.com>;tag=as1ba1c908' If I change the Dial entry in extension.conf to exten => _.,3,Dial(SIP/${EXTEN}@fwd.pulver.com,20) Then I no longer see the outgoing connection being handled as if it is going across a NAT. The call does connect but there is only sound from the asterisk originatin...

....0.2, requested format = 2, actu al format = 2 -- Executing SetCallerID("IAX[2203@2203]/9", "91184") in new stack -- Executing SetCIDName("IAX[2203@2203]/9", "calisto") in new stack -- Executing Dial("IAX[2203@2203]/9", "SIP/612@fwd.pulver.com") in new stack We're at 82.38.193.149 port 16612 Answering with preferred capability 4 Answering with preferred capability 2 Answering with non-codec capability 1 11 headers, 10 lines Reliably Transmitting: INVITE sip:612@fwd.pulver.com SIP/2.0 Via: SIP/2.0/UDP 82.38.193.149:5060;branc...

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 username=74573 host=fwd.pulver....

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)" back from 69.90.155.70 -- SIP/fwd.pulver.com-f526 is circuit-busy == Everyone is busy/congested at this time Have you any idea? Thank y...

..._________ From: Carl Ford [mailto:carlf at vonmag.com] Sent: Wednesday, 18 July 2007 9:09 AM To: Dean Collins Subject: Announcing Digium/Asterisk World's Conference Program <http://www.magnet101.com/ls.cfm?r=49686240&sid=2533069&m=336237&u=Pulve rmedi&s=https://secure.pulver.com/digiumAsteriskWorld/2007/boston/web/at tendRegister.htm> Hi Dean, I am pleased to announce that the conference program <http://www.magnet101.com/ls.cfm?r=49686240&sid=2533070&m=336237&u=Pulve rmedi&s=http://www.digiumasteriskworld.com/2007/boston/web/confSchedule. htm&...

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 other one, &quot...

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.pulv...

...CONF FILES WORKING WILL BE APPRECIATED THANKS! server*CLI> sip show registry Host Username Refresh State 69.90.155.70:5060 431044 160 Registered 69.90.155.70:5060 421058 160 Registered SIP.conf register => 421058:password@fwd.pulver.com/103 ;Register Free World Dialup register => 431044:password@fwd.pulver.com/103 [fwd1] type=friend username=431044 secret=password fromuser=431044 fromdomain=fwd.pulver.com host=fwd.pulver.com insecure=very canrenvite=no nat = yes dtmfmode=inband [fwd2] type=friend secret=password username=...

Same thing. It will make sense to try Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwdnat.pulver.com:5082 but in that case Asterisk sends REGISTER sip:fwdnat.pulver.com SIP/2.0 which is not right. It should be sip:fwd.pulver.com but sent thru fwdnat.pulver.com:5082 BR Borut -----Original Message----- Subject: Re: [Asterisk-Users] Using Asterisk with FWD...

...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 behind a NAT. It can be found on: http://www.pulver.com/fwd/index.html Anyway, I set up an acount and as long as my * box dials into the internet itself it works fine. But as soon as I try to connect from behind the NAT I can't here the other person (he can here me though). My setup: ATA -> * -> NAT -> Freeworld -> other person Us...

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 IP" message. In second scenario I hack hosts file that fwd.pulver.com points to fwdnat.pulver.com. In that case I w...

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 mailbox: exten => 700001,1,Dial(SI...

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, My configuration: Sipura 2000 Debian/Sarge Asterisk 1.0.1 built by msp@toshiba on a i686 running Linux I am calling 612@fwd.pulver.com which is Daytimephoneline of pulver.com and for the first second the connection seems to be ok and I hear: <thu .. rsda ..> and nothing more, which suppose to mean <Thursday, ...>. The echo line from 613@fwd.pulver.com does work the same. I never got an echo back, the connection see...

...ble to integrate it into asterisk? I don't konw how to program so I'm really sorry I can't do it myself. But maybe some of you more skilled programmers can make this possible ;-). I think the possibilities would be great. There are more and more services like freeworlddialup (http://www.pulver.com/fwd/index.html) that could help voip to break trough for private users and I'd just hate not to be able to participate just because I'm behind a nat and using asterisk when other people with a regular sip client don't have this problem from behind a nat. If anybody could help me ac...

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 from a SIP standpoint. I can dial FWD numbers like 612/613/etc and everythin...

FWD (Free World Dialup) allows any SIP call to US toll free numbers via * 18xxzzzyyyy at fwd.pulver.com This works WITHOUT the need to be registered at FWD so in my dialplan I have something like: exten => _8.,1,Dial(SIP/fwd.pulver.com/*${EXTEN:1},60,r) exten => _8.,2,Hangup And I just dial 8-1-8xxyyyzzzz and presto ... calls go through just fine 99% of the time. I'm wondering if...

...network (ie, FWD 170099XXXXX gatewayed numbers work). To get FWD, IPTel, or SIPPhone to work, I've deduced I need to do something like the following after reading various online email archives (please correct me if I'm wrong): sip.conf: [general] register => XXXXX:password@fwd.pulver.com/1000 register => 1747XXXXXXX:password@proxy01.sipphone.com/1000 register => username:password@iptel.org/1000 extensions.conf: [default] include=sip [sip] include=sip [fwd] exten => _91339.,1,SetCallerID(XXXXX) exten => _91339.,2,Dial,SIP/${EXTEN:1}@fwd.pulver.c...

I recently got a Pulver Innovations WiSIP wireless SIP phone to determine if we want to use them in our organization. Since the WiSIP phone arrived last week, I have had nothing but headaches. I do think I now have the problem narrowed down. I have spent a bulk of my time trying to get the WiSIP to work with a couple o...