Since you've specified that the gs102 peer has a dynamic IP address, you'll need to ensure that this peer registers with Asterisk, otherwise it'll default to the 192.168.2.1 address in the config file.
ast guy wrote: > Will it require to add register statement in sip.conf. I have all sip > buddies in Database. so will that work in this scenario ? > -ag > > On Feb 10, 2008 11:55 AM, Rob Hillis <[EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]>> wrote: > > Why are you specifying the password and server IP in the dial > string when it's included in sip.conf? It's unnecessary. > > I believe that Dial(SIP/gs102/1234) will achieve what you want. > > ast guy wrote: >> Hi, >> >> I'm trying to call a SIP server while providing the SIP server >> username/password in dial string but it's not working ... >> >> Dial(SIP/gs102:[EMAIL PROTECTED] >> <mailto:SIP/gs102:[EMAIL PROTECTED]>); >> >> User on sip server (192.168.2.81 <http://192.168.2.81>): >> >> [gs102] >> disallow=all >> allow=ulaw >> allow=alaw >> type=friend >> username=gs102 >> secret=test >> host=dynamic >> dtmfmode=inband >> defaultip=192.168.2.1 <http://192.168.2.1> >> qualify=1000 >> mailbox=102 >> context=context-gs102 >> >> Extensions.conf entry >> >> [context-gs102] >> >> exten => s,1, Answer(); >> exten => s,n, Playback(demo-congrats); >> exten => s,n, Meetme(8600051); >> >> exten => 1234,1, Answer(); >> exten => 1234,n, Playback(demo-congrats); >> exten => 1234,n, Meetme(8600051); >> >> >> When I dial I get following error on console >> >> -- Executing Dial("SIP/331-6263", "SIP/gs102:[EMAIL PROTECTED] >> <mailto:SIP/gs102:[EMAIL PROTECTED]>") in new stack >> -- Called gs102:[EMAIL PROTECTED] >> <mailto:gs102:[EMAIL PROTECTED]> >> -- SIP/192.168.2.81-0343 is circuit-busy >> == Everyone is busy/congested at this time (1:0/1/0) >> -- Executing Hangup("SIP/331-6263", "") in new stack >> == Spawn extension (default, 1234, 2) exited non-zero on >> 'SIP/331-6263' >> >> >> I want to call extension 1234 defined under gs102 defined >> context-gs102 context... what should be the exact Dialed SIP URL ? >> >> >> -ag >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
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