[Asterisk-Users] Working with FWD, IPTel, SIPPhone?

[Eric Wieling]

These list messages might be useful:

http://lists.digium.com/pipermail/asterisk-users/2003-June/013918.html
http://lists.digium.com/pipermail/asterisk-users/2003-June/013946.html

On Tue, 2003-08-12 at 13:22, 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: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Ian Blenke
> Sent: Tuesday, August 12, 2003 12:06 PM
> To: asterisk-users at lists.digium.com
> 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:password at fwd.pulver.com/1000
> 	 register => 1747XXXXXXX:password at proxy01.sipphone.com/1000
> 	 register => username:password at iptel.org/1000
> 
> 	extensions.conf:
> 	 [default]
> 	 include=sip
> 
> 	 [sip]
> 	 include=sip
> 
> 	 [fwd]
> 	 exten => _91339.,1,SetCallerID(XXXXX)
> 	 exten => _91339.,2,Dial,SIP/${EXTEN:1}@fwd.pulver.com,tr
> 
> 	 exten => _91747.,1,SetCallerID(1747XXXXXXX)
> 	 exten => _91747.,2,Dial,SIP/${EXTEN:1}@proxy01.sipphone.com,tr
> 
> 	 exten => _91478.,1,SetCallerID(XXXXXXXX)
> 	 exten => _91478.,2,Dial,SIP/${EXTEN:1}@iptel.org,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,
-- 
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