similar to: SIP connection

Hi, I've posted a simular message little over a week ago so sorry for reposting. I need to register to freeworld dial up from behind a nat. Using the xten software sip client works fine but with asterisk I don't know how to do it. Last time I posted I got different responses. Some saying I can't register with an outbound proxy from asterisk others said they have done it. If it is

Could someone please point me to the proper url to register for a fwd acount and get a fwd number. I couldn't find it at www.freeworlddialup.com or fwd.pulver.com Thanks, -Galt -- "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin Franklin

Hello, I am trying to setup my Asterisk box to accept calls from the FWD network. I've followed all the config advice / samples I've found on the web. Making calls to devices on the FWD network from my Asterisk box works flawlessly, but whenever I try to call my Asterisk box from a FWD client I get a busy signal, and a "Call Disconnected" 486 error. What's odd is that I

There is now a peering arrangement between e164.org and FreeWorldDialup which means any and all subscribers on FWD are now easily able to make enum calls by prefixing their call with **164, like wise it's almost as simple to make a call to FWD by hitting 8829990<fwd number> This means that for those of you wanting to send/receive calls to/from FWD subscribers you can now do so, easily

I'v a sip client and a sip trunk to FWD: [general] port=5060 context=default tos=reliability disallow=all allow=ulaw careinvite=no [freeworlddialup] context=default type=friend username=MYUSERNAME secret=MYPASSWORD host=fwd.pulver.com [igor] type=friend callerid="Me" host=dynamic dtmfmode=rfc2833 careinvite=no When i try to call a FWD number from SIP client i obtain a lot of

I'm trying to test IAX2 with FWD It registers fine but when I try to receive the call I get: chan_iax2.c:476 iax_error_output: Ignoring unknown information element 'Unknown IE' (38) of length 1 Jan 25 18:02:12 WARNING[114696]: chan_iax2.c:476 iax_error_output: Ignoring unknown information element 'Unknown IE' (39) of length 1 Jan 25 18:02:12 WARNING[114696]: chan_iax2.c:476

Hello, I have 2 channels in iax.conf [iaxfwd] type=user callerid= Free World Dialup inkeys=freeworlddialup auth=rsa context=incoming qualify=yes [iaxfwd-outbound] type=peer host=iax2.fwdnet.net username=xxxxxx secret=*********** auth=md5 The problem is: When I tell FWD to call me I have this output in my asterisk consol: Executing Dial("IAX2/iaxfwd-outbound-3",

Hi, My asterisk server is behind firewall and i am trying to connect to FWD. i hv configured as mentioned in this link http://www.freeworlddialup.com/advanced/iax. i am able to register my server with FWD. But when i dial 393612, i always get 'No one is available to answer this time, try again later'. I hv portforwarded tcp 4569 and 5060 from my firewall to my asterisk server. Any idea

Hello All, I have a problem. I have tried everything that is in the book "The Future of Telephony" as well as on the FWD (freeworlddialup) website, and there is still a problem. My asterisk box is not able to associate with the FWD server. I get: Registration Rejected by [insert IP], and I can't use my IPCall number to reach my Asterisk box. Any suggestions? -- -Shane Blog:

Hello, Sorry for what may be a basic question, but I have spent a number of hours trawling the forums without resolving the problem, and hence this post. I have just started to dabble with Asterisk, as much for the learning than anything else. I created an account on FWD and used the Asterisk settings that the FWD web site recommends at

When I try to call out to FWD over IAX2 I get: Call rejected by 65.39.205.121: Unable to negotiate codec I'm using asterisk-1.0.5 (the same settings works fine with *0.9) I've standard settings in iax.conf [general] bindport=4569 register => xxxxx:xxxxxx@iax2.fwdnet.net [iaxfwd] type=user context=fromiaxfwd auth=rsa inkeys=freeworlddialup disallow=all allow=ulaw -- #Joseph

============= SJphone Log ============ Outgoing SIP session Respondent: (sip:8612@192.168.2.2) Remote client: Started: May 26 16:33 Accepted: no Ended: May 26 16:34 End reason: Call rejected: 503 Service Unavailable =============== Asterisk Debug ================ Executing Dial("SIP/2201-a83e", "IAX2/<FWDNUMBER>:@iax2.fwdnet.net/612|60|r") in new stack --

I have configured iax.conf and extensions.conf as instructed on FWD website (http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76) and I can successfully receive calls and make test calls to 612, 613, etc. The problem is that I can not make a call to another FWD user. Here is what asterisk says: -- Executing [393xxxxxx@default:1] Set("Zap/1-1",

Hello !! I saw in FWD site a phone on the web.. (click 612 link) http://www.freeworlddialup.com/advanced/beta_programs I?d like to have this application in my intranet.. click on my name, than calls my number.. I?d also like to see that phone on the web... as an option How can I do that ? Is it possible to download ? Any related link ? Thanks Andrei.

Hi Folks, Just wondering if anyone can give me some pointers, I'm configuring Asterisk to talk to FWD's new IAX service. The asterisk server is behind an iptables NAT Firewall, with port 5036 forwarded: $IPTABLES -t nat -A PREROUTING -p udp -d $EXTERNAL_IP --dport 5036 -j DNAT --to-destination 172.16.20.200:5036 I can make outgoing calls just fine, but when I receive an inbound call

Good afternoon: I need if they can help me I have 2 Servants with Fedora Core 1 Kernel-2.4.22-1.2197.ntpl, with the following packages Openswan: Kernel-modulate-openswan-2.4.22-1.2197.ntpl-2.1.4-fc1.dag.i686.rpm and the Openswan-utils-2.1.4-1.fc1.dag.i686.rpm. As Firewall I have the shorewall 2.1 All this works well when having alone an In-date IP. When having 2 In-date IP, one to leave to

Hello all, Note: This is still an asterisk-based discussion, but we widened the interests of the conference and also wish to avoid trademark abuse, although we continue work closely with Digium,hence the change of name. Today, we'll be listening to the latest on Astricon which is fast approaching. Our featured guest is Daniel Berninger from FWD. FreeWorldDialup.com has brought a lot of

I have an asterisk server running on Redhat 8.0 with a Digium TDM400P w/4 FXO modules (TDM04P) There are 2 lines going into the Digium card. One line is a Vonage digital line, and the other line is a Comcast voice line. I have a SIP Grandstream 100 phone connected to the Asterisk server. I also have IAX configured with FWD. The problem is that on occasionally, after talking for about 20

Good morning all, I'm setting up Asterisk for the first time with no prior PBX experience. I'm following Andy Powell's 'Getting Started with Asterisk' (http://www.automated.it/guidetoasterisk.htm). This is my second time through that document - as I did something weird the first time and really upset it somehow - and I wanted to ask a few general questions of the list.

Hello all, I am trying to make a little bit more complex QoS/Shapping form and I need to shape a PPPoE conection that I serve to my clients. So this is the scope: client connects using pppoe so it gets an IP address (from pppoe pool) and open an interface into my linux box interface for this client is ppp0 client has got an ip 1.1.1.2/32 and it is poiting to pppoe-server 1.1.1.1 So,