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]
