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