One of my boxes uses a public IP address, the other one is behind a NAT router. The second one is not setup with DMZ or any port forwarding. However, I am using STUN. Something like this:
[general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes defaultexpirey=3000 stunaddr=some.stun.server.com externrefresh=180 useragent=asterisk rtpkeepalive=10 register => 1647xxxx...@fpl_peer [fpl_peer] type=friend username=1647xxxxxxx secret=xxxxxxxx host=voip.freephoneline.ca insecure=port,invite nat=yes qualify=no allow=ulaw canreinvite=no On Fri, Feb 12, 2010 at 4:33 PM, terry D. Cudney <[email protected]> wrote: > Hi Liviu, > > Thanks for the config... I actually had found it from your previous post > today and adjusted my sip settings to match what you have. > > This is asterisk v 1.4.21.2. You said that you have it running on a 1.4 > box, so I presume that it isn't a difference between 1.4 - 1.6. > > The problem may be in my nat setup. I'm grasping at straws here... The > asterisk box is an atom-based machine sitting in the DMZ behind a linksys > wrt54gl running openwrt v0.9. > > When I do: sip set debug host fpl from the cli, I get output that seems to > me, to show that my box and the fpl server are talking ok. I'm attaching a > screen capture of the dialog after I "sip set debug peer fpl". > > I haven't figured out how to capture the output when I make a call, but > when I do make a call, the output shows the SIP response 603 from the fpl > server (scrolls off screen) and I get the audio message on the phone, "This > call is not covered by your tariff plan" or something similar. > > I'm out of ideas on where to look for the answer here. Any other > suggestions on how to trace what is happening/why the call is being rejected? > > I'd appreciate any suggestions you can make. Or, if you have the time, you > can call me at SIP:[email protected], ext 1. > > Thanks very much, > > --terry > > On Fri, Feb 12, 2010 at 02:22:08PM -0500, Liviu Toma wrote: >> This is what works for me: the useragent is set to just "asterisk" (no >> quotes) and the entries in sip.conf are like below. Asterisk 1.6.2.1 >> right now, but I've been using it since 1.6.0.x >> >> register => 1416xxxx...@fpl_peer >> >> [fpl_peer] >> type=friend >> context=default ; the default context for incoming calls >> username=1416xxxxxxx >> secret=xxxxxxxx >> host=voip.freephoneline.ca >> insecure=port,invite >> nat=no >> qualify=no >> allow=ulaw >> allow=g729 >> canreinvite=no >> >> I also have it running on an Asterisk 1.4, on a Linksys NSLU2 >> >> Liviu >> >> On Fri, Feb 12, 2010 at 1:11 PM, Peng Li <[email protected]> wrote: >> > HI, >> > >> > Please refer to this link. >> > http://forum.freephoneline.ca/viewtopic.php?f=14&t=678&start=25 >> > >> > The only thing changed is that we have to use useragent=xxxxx in the >> > general >> > section of SIP.conf. The rest will stay same. >> > >> > Peng >> > >> > >> > On Fri, Feb 12, 2010 at 11:01 AM, terry D. Cudney >> > <[email protected]>wrote: >> > >> >> Hi guys, >> >> >> >> Thanks to saurin, Bruce and Reza for the suggestions on getting >> >> freephoneline to work... >> >> >> >> saurin, the codecs are g711u and g729... that doesn't seem to be the >> >> trouble spot. >> >> >> >> Bruce, thanks for the suggestion re UserAgent in sip.conf. That does >> >> change the response I get from freephoneline.ca. Setting the >> >> useragent=freephoneline eliminates the SIP response 603. But then the >> >> audio >> >> response on making an outbound call is that the number called is not >> >> covered >> >> by the tariff plan.... which is ridiculous since the call was to a number >> >> local to the freephoneline DID It DOES appear that freephoneline is >> >> disallowing useragent=Asterisk PBX. <growl> >> >> >> >> Reza, thanks for the pointer to the SIP RFC. That will be very useful in >> >> future too! >> >> >> >> In summary, re freephoneline.ca: My asterisk v1.4.21.2 (not one of the >> >> GUI versions) registers successfully with voip.freephoneline.ca and >> >> accepts/completes incoming calls. However, outgoing calls do not work so >> >> far. With useragent=Asterisk PBX, I get the SIP response 603. With >> >> useragent=freephoneline, on dialing a number local to the fpl DID I get >> >> the >> >> audio message: That number is not covered by your tariff plan. >> >> >> >> I believe some people on this list, are using fpl successfully. Peng, I >> >> lost your email address. The sip.conf and extensions.conf snippits that >> >> you >> >> sent me don't work here... Is there something else you have set >> >> differently? >> >> Or, >> >> >> >> If anyone is using freephoneline.ca successfully would you be willing to >> >> share your information on how you got it working? >> >> >> >> Thanks again to all, >> >> >> >> --terry >> >> >> >> On Fri, Feb 12, 2010 at 03:52:03AM -0500, Reza - Asterisk Consultant >> >> wrote: >> >> > http://www.rfc-ref.org/RFC-TEXTS/3261/chapter21.html >> >> > >> >> > 603 Decline >> >> > >> >> > The callee's machine was successfully contacted but the user >> >> > explicitly does not wish to or cannot participate. The response MAY >> >> > indicate a better time to call in the Retry-After header field. This >> >> > status response is returned only if the client knows that no other >> >> > end point will answer the request. >> >> > >> >> >> >> -- >> >> >> >> Name: Terry D. Cudney >> >> Phone: (705) 812-6744 >> >> E-mail: [email protected] >> >> SIP: [email protected] >> >> >> >> --------------------------------------------------------------------- >> >> To unsubscribe, e-mail: [email protected] >> >> For additional commands, e-mail: [email protected] >> >> >> >> >> > >> >> --------------------------------------------------------------------- >> To unsubscribe, e-mail: [email protected] >> For additional commands, e-mail: [email protected] > > -- > > Name: Terry D. Cudney > Phone: (705) 812-6744 > E-mail: [email protected] > SIP: [email protected] --------------------------------------------------------------------- To unsubscribe, e-mail: [email protected] For additional commands, e-mail: [email protected]
