[Asterisk-Users] Working with FWD, IPTel, SIPPhone?
Brian West
brian at bkw.org
Tue Aug 12 13:16:12 MST 2003
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: 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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