i have an asterisk box (195.112.214.99) with this configuration: sip.conf [callshop] type=peer host=sip.callshopcompany.com username=XXXXXXX secret=XXXXXX allow=all
extensions.conf [call] exten => _00.,1,Dial,SIP/callshop/${EXTEN} and when i try to send calls to the voip provider (callshopcompany "213.61.187.150") i got these messages: *CLI> dial [EMAIL PROTECTED] -- Executing Dial("OSS/dsp", "SIP/callshop/0017046872001") in new stack -- Called callshop/0017046872001 *CLI> Sep 24 14:16:45 WARNING[22295]: chan_sip.c:6890 handle_response: Forbidden - wrong password on authentication for INVITE to '"asterisk" <sip:[EMAIL PROTECTED]:5070>;tag=as4cda63c2' -- SIP/callshop-f613 is circuit-busy == Everyone is busy/congested at this time -- Got SIP response 481 "Call Leg Does Not Exist" back from 213.61.187.150 Sep 24 14:16:58 WARNING[22295]: pbx.c:1949 ast_pbx_run: Timeout, but no rule 't' in context 'call' << Hangup on console >> but when ive tried it on xlite in the same configuration to send calls to the same company it worked and the calls passed without any problems. so whats the problem here,why the call goes well using xlite and fails using asterisk despite they have the same configuration. --- Rich Adamson <[EMAIL PROTECTED]> wrote: > > > i tried to send calls through an asterisk box to a > > voip provider the calls failed and here what i got > : > > > > *CLI> Sep 24 11:09:19 WARNING[23356]: > chan_sip.c:6890 > > handle_response: Forbidden - wrong password on > > authentication for INVITE to '"asterisk" > > <sip:[EMAIL PROTECTED]:5070>;tag=as667cb0ae' > > -- SIP/call-3f73 is circuit-busy > > == Everyone is busy/congested at this time > > -- Got SIP response 481 "Call Leg Does Not > Exist" > > back from 213.61.187.150 > > > > but when i have tried to send calls using xlite > > softphone it worked and the calls passed without > any > > problems. > > You've made a hell of a lot of assumptions that we > understand > your configuration, and we don't. > > What is 195.112.214.99 and 213.61.187.150? > > Is your sip phone registered with asterisk? (what > does sip show > peers indicate?) > > Is your sip phone or asterisk registering with your > sip provider? > (what does sip show registry indicate?) > > Paste the appropriate sections of sip.conf and > extensions.conf > along with some clue what addresses and extensions > are what. > > > _______________________________________________ > --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 > __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com _______________________________________________ --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