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-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shane D Sent: Monday, January 07, 2008 19:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FWD and IPCall no-ip.org appears to want to charge me money... Is there a free alternative? On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote: > > If you want to forward your ipkall number directly to your asterisk > server: > > 1. If your asterisk server is on a private LAN and is connected to the > internet via a router, enable the router to port forward UDP/5060 & > UDP/10000-20000 to your asterisk server (assuming you have not changed > rtp config parameters in rtp.conf). > > 2. Check that the firewall (if any) on your asterisk server allows > connections on UDP/5060 & UDP/10000-20000 > > 3a. Static public IP address - use the fully qualified domain name > assigned to the IP address (or setup an account on www.no-ip.org with a > name of your choice) > > 3b. Dynamic public IP address - setup an account on www.no-ip.org with a > name of your choice - install the dynamic ip address update client to > monitor any change of your ip address (downloads & instructions on > no-ip.org website) > > 4. Goto www.ipkall.com and login to your account. Use your ipkall number > as the SIP Phone Number and then the name you selected in 3a or 3b as > the SIP Proxy. > > 5. Wait 60 minutes for changes to take affect (!) > > 6. Edit asterisk sip configuration to allow calls from ipkall: > > vi /etc/asterisk/sip.conf and find the section beginning [general] > > Add/replace the following: > > externhost=the name you setup in 3a. or 3b. > localnet=your private LAN e.g. 192.168.2.0/255.255.255.0 > > Add a new section at the bottom of the file: > > [ipkall.com] > host=voiper.ipkall.com > context=from-ipkall > dtmfmode=rfc2833 > insecure=invite > type=friend > canreinvite=no > disallow=all > allow=ulaw ; you can add other codecs if you want once the setup works > > Save the file. The section you added tells asterisk to accept calls from > voiper.ipkall.com and to place them in the "from-ipkall" context. This > context can be whatever you want. You may need to change the insecure= > line if you are using asterisk 1.2 > > 7. Edit asterisk dialplan configuration to handle calls from ipkall: > > vi /etc/asterisk/extensions.conf and add at the bottom: > > [from-ipkall] > exten => <IPKALL-NUMBER>,1,NoOp(from-ipkall) > exten => <IPKALL-NUMBER>,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM}) > exten => <IPKALL-NUMBER>,3,Dial(Local/[EMAIL PROTECTED]) > > Save the file. The section you added tells asterisk what to do with > calls that are received in the "from-ipkall" context. Replace the > <IPKALL-NUMBER> with whatever you entered in the SIP Phone number field > on the ipkall website (I recommended your ipkall number). > > In the "from-ipkall" section: > 1: display "from-ipkall" on the console > 2: display the caller id & name > 3. phone the local extension 200 in context "local" - replace this line > with your personal requirements. > > Connect to the asterisk console (asterisk -R on my server) and "sip > reload" followed by "dialplan reload" (asterisk 1.4) or "extensions > reload" (asterisk 1.2). "sip reload" will re-read the sip.conf file & > "dialplan reload"/"extensions reload" will re-read the extensions.conf > file. > > Phone your ipkall number and see if anything is displayed on the console > and/or your phone rings. > > If nothing on the console when you phone, try "sip set debug peer > ipkall.com" (asterisk 1.4 - not sure of the command for asterisk 1.2) > and phone again. > > Post back your results. > > > > > > > > > > > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Shane D > Sent: Monday, January 07, 2008 17:32 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] FWD and IPCall > > Okay... That was kind of confusing. Would you contact me off-list to > help me specifically? > > I've double-checked everything for the IAX, and it's a no-go. Maybe > I'll try this SIP thing. But then again, if I can just hook IPKall to > the server directly, I don't need FWD... > > On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote: > > My config is as follows > > > > Excerpt of sip.conf: > > > > [general] > > externhost=fully.qualified.domain.name > > localnet=192.168.2.0/255.255.255.0 > > srvlookup=no > > defaultexpiry=3600 > > dtmfmode=rfc2833 > > > > register => <fwd-id>:<fwd-pwd>@fwd.pulver.com/<fwd-id> > > > > [sipfwd] > > type=peer > > secret=<fwd-pwd> > > username=<fwd-id> > > fromdomain=fwd.pulver.com > > host=fwd.pulver.com > > disallow=all > > allow=ulaw > > canreinvite=yes > > insecure=invite > > qualify=yes > > context=from-fwd > > > > Excerpt of extensions.conf: > > > > [from-fwd] > > exten => <fwd-id>,1,NoOp(from-fwd) > > exten => <fwd-id>,n,Dial(whatever) > > > > I have a dynamic public IP address, so I use http://www.no-ip.org to > map > > my IP address to name. My router port forwards UDP/5060 & > > UDP/10000-20000 to the internal asterisk server. > > > > However, I do not have ipkall forwarding to my fwd account. I have it > > forwarding directly to my asterisk server using the no-ip.org address > I > > registered. > > > > e.g. forward to sip:[EMAIL PROTECTED] on ipkall website > > and then in sip.conf: > > > > [ipkall.com] > > host=voiper.ipkall.com > > context=from-ipkall > > dtmfmode=rfc2833 > > insecure=invite > > type=friend > > canreinvite=no > > disallow=all > > allow=ulaw > > > > And in extensions.conf: > > > > [from-ipkall] > > exten => xxx,1,NoOp(from-ipkall) > > exten => xxx,n,Dial(whatever) > > > > > > > > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Shane D > > Sent: Monday, January 07, 2008 12:09 > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] FWD and IPCall > > > > It's Iax2. Is there a way of using amore reliable sip > > connectoin/something slightly different? > > > > If so, how would I go about that. > > > > On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote: > > > You haven't said if your connection to fwd is SIP or IAX2 but I have > > > found IAX2 connections to fwd to be unreliable. Other people may > have > > > different results. > > > > > > -----Original Message----- > > > From: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] On Behalf Of Shane > D > > > Sent: Monday, January 07, 2008 10:17 > > > To: asterisk-users@lists.digium.com > > > Subject: [asterisk-users] FWD and IPCall > > > > > > 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: http://blind-geek.com/blog/ > > > CoOwner: http://sjtechzone.com > > > AIM: inhaddict > > > Skype: chatter8712 > > > > > > _______________________________________________ > > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > -- > > -Shane > > Blog: http://blind-geek.com/blog/ > > CoOwner: http://sjtechzone.com > > AIM: inhaddict > > Skype: chatter8712 > > > > _______________________________________________ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > -Shane > Blog: http://blind-geek.com/blog/ > CoOwner: http://sjtechzone.com > AIM: inhaddict > Skype: chatter8712 > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- -Shane Blog: http://blind-geek.com/blog/ CoOwner: http://sjtechzone.com AIM: inhaddict Skype: chatter8712 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users