While we are on this subject. For testing and such I have been trying to get one asterisk server to register with another via sip(i know i know use IAX) but it doesn't work.... It should... I can't see any reason it shouldn't.... Any pointers? All I get is proxy auth and * crashes a bloody death on the client side.
bkw On Tue, 12 Aug 2003, Steve Lane wrote: > I am trying to do the same thing you are doing. I am new to asterisk and > a friend of mine owns a carrier. They are using vocal data as the > platform, which is sip capable and uses sip phones. What I was trying to > do as well is register * with the redirect/registers with the carrier so > that they can route my outbound calls outside of the LAN. All internal > calls would remain the responsibility of Asterisk. Is this possibly the > same thing you are trying to accomplish? > > Steve > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Ian Blenke > Sent: Tuesday, August 12, 2003 12:06 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Working with FWD, IPTel, SIPPhone? > > 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 to work with IAXTEL wasn't a problem, but I'm still fighting > > with inbound/outbound VoIP "trunks" with IAX2 or anything SIP. I can > call 1700NXXNXXX IAXTEL numbers, and anything gatewayed from that > 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:[EMAIL PROTECTED]/1000 > register => 1747XXXXXXX:[EMAIL PROTECTED]/1000 > register => username:[EMAIL PROTECTED]/1000 > > extensions.conf: > [default] > include=sip > > [sip] > include=sip > > [fwd] > exten => _91339.,1,SetCallerID(XXXXX) > exten => _91339.,2,Dial,SIP/${EXTEN:[EMAIL PROTECTED],tr > > exten => _91747.,1,SetCallerID(1747XXXXXXX) > exten => _91747.,2,Dial,SIP/${EXTEN:[EMAIL PROTECTED],tr > > exten => _91478.,1,SetCallerID(XXXXXXXX) > exten => _91478.,2,Dial,SIP/${EXTEN:[EMAIL PROTECTED],tr > > Unfortunately, this doesn't appear to work. Nor do any other > translations (even a simple "_8." doesn't work). No matter what I try, I > > keep getting "404 Not found" or "all circuits are busy" messages. > > As far as I can tell, I'm registered with with all three SIP providers: > > *CLI> sip show registry > 195.37.77.101:5060 username 120 Registered > 192.246.69.223:5060 XXXXX 120 Registered > 130.94.123.252:5060 1747XXXXXX 120 Registered > > I'm also apparently registered correctly with IAX and IAX2: > > *CLI> iax show registry > Host Username Perceived Refresh State > 12.37.165.130:5036 username 66.x.x.x:5036 60 Registered > > *CLI> iax2 show registry > Host Username Perceived Refresh State > 12.37.165.130:4569 username 66.x.x.x:4569 60 Registered > > Unfortunately(?), any calls through IAX2 never seem to go through. > > While I'd like to eventually setup an outbound NAT proxy, I've had a > difficult time decyphering how to configure SER, siproxd, or PartySIP to > > register to external SIP providers like FWD, IPTel, and SIPPhone. I'm > guessing this is what the additional sections in sip.conf are for? > > sip.conf > > ;; Free World Dialup Proxy > [fwd.pulver.com] > type=friend > host=fwd.pulver.com > fromuser=48702 > fromdomain=fwd.pulver.com > ;secret=password > ;username=XXXXX > > Do you need these sections if you're not NATting? How would I define > fwdnat.pulver.com:5082 above? (asterisk appears to treat the whole > string as a hostname). > > At some point, I'd like to have branch offices off of IPSEC tunnelled > connections - running an Asterisk instance on every customer's firewall > isn't as appealing as a simple SIP proxy. > > I guess the confusion is: how do you setup a SIP Provider *and* an > outbound proxy (either locally on my linux firewall, or provided by the > SIP carrier?) > > This really could use a good HOWTO/FAQ, but for the life of me I can't > find it (if someone would take the time to guide me a bit with this, I > wouldn't mind taking a stab at writing one). > > Thanks, > > -- > - Ian C. Blenke <[EMAIL PROTECTED]> > (This message bound by the following: > http://www.nks.net/email_disclaimer.html) > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
