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

[Steve Lane]

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,

-- 
- Ian C. Blenke <icblenke at nks.net>
(This message bound by the following:
http://www.nks.net/email_disclaimer.html)


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