I have a problem. Incoming calls work without problem but I cant call out. Using AAH.Gets a busy tone Anyone who can see a mistake in Outgoing settings context=from-pstn host=ipkund1.rixtelecom.se insecure=very nat=yes secret=xxxxxxxxxxx type=peer username=0406082250 Regards Anders Svensson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051004/71f64548/attachment.htm
Hi, Outgoing setting is in zapata.conf. I think you should read the wiki more ;). If what you mean by outgoing is another sip extension then you should look for extension.conf. Links: http://www.voip-info.org/wiki-Asterisk+config+zapata.conf http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf Regards, Stevanus Anders Svensson wrote:> I have a problem. Incoming calls work without problem but I cant call > out. Using AAH.Gets a busy tone > > Anyone who can see a mistake in Outgoing settings > > > > > context=from-pstn > host=ipkund1.rixtelecom.se > insecure=very > nat=yes > secret=xxxxxxxxxxx > type=peer > username=0406082250 > > > > Regards > > Anders Svensson > > > >------------------------------------------------------------------------ > >_______________________________________________ >--Bandwidth and Colocation sponsored by Easynews.com -- > >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051004/3d5b70f3/attachment.htm
Anders Svensson wrote:> I have a problem. Incoming calls work without problem but I cant call > out. Using AAH.Gets a busy tone > > Anyone who can see a mistake in Outgoing settings > > > > > context=from-pstn > host=ipkund1.rixtelecom.se > insecure=very > nat=yes > secret=xxxxxxxxxxx > type=peer > username=0406082250 >"username" is one of the most misunderstood settings in sip.conf and it's really a bad, bad, bad name. You want to set fromuser and fromdomain together with username. "username" has many uses, which is bad: * One is to set a default user name that is used in combination with default IP when we have no registration from a local peer. * The other use is what you are trying to set up: to set authentication username when we register and place calls to an outbound service provider. This is always used in combination with "fromuser" and "fromdomain". The nat=yes setting seems redundant, it should be done on Rix telecom's side. When you set nat=yes you tell asterisk that the *other end* is behind nat. I do not believe a service provider run a service behind NAT. Good luck! Lycka till! /Olle
stevanus wrote:> Hi, > > Outgoing setting is in zapata.conf. I think you should read the wiki > more ;). > If what you mean by outgoing is another sip extension then you should > look for extension.conf. > > Links: > http://www.voip-info.org/wiki-Asterisk+config+zapata.conf > http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf >And in fact it was all about sip.conf ;-) As you say, connecting to SIP service providers is well documented on the wiki, but not on those pages. /O