You're attempting to connect on ports 5061-5062 but are bound to port 5060...?
What does your CLI look like during a failed call attempt? Thanks, --Warren Selby On Fri, Oct 30, 2009 at 2:18 PM, Cliconnect <[email protected]>wrote: > > > Thank you, > > > >>How are you setting up xlite and the ata? > > Xlite > > User name : 1000 > Domain: IP of the server running Asterisk > Register with domain and receive incoming calls: clear > Port used in local computer : manually specify range : 5061-5062 > > ATA > SIP server address: IP of the server running Asterisk > Outbond Proxy : IP of the server running Asterisk > SIP User id : 1001 > Accoount ID : 1001 > Use DNS SRV : yes > User id is phone number : yes > SIP registration : no > Local sip port : 5062 > > > >>Which version of Asterisk are you using? > Asterisk 1.6.1.6, Copyright (C) 1999 - 2009 Digium, > > > >> What does the general section of your sip.conf look like? > > [general] > context=default > allowoverlap=no > bindport=5060 > bindaddr=0.0.0.0 > srvlookup=yes > > When I > > sip show peers > > Name/username Host Dyn Nat ACL Port Status > 1000 (Unspecified) D 5060 Unmonitored > > 1001 (Unspecified) D 5060 Unmonitored > > 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 > offline] > > regards > > Jair Santos > > > > > > > Warren Selby wrote: > > How are you setting up xlite and the ata? Which version of Asterisk are > you using? What does the general section of your sip.conf look like? > > On Fri, Oct 30, 2009 at 1:01 PM, Cliconnect <[email protected]>wrote: > >> Hi all, >> >> I can only get a line signal when I set the phones to not register with >> domain . >> >> All phones are in the same NAT and I cannot make calls. >> >> I am getting "Call failed : Proxy Authentication Required" in Xlite and a >> busy signal when using an ATA. >> >> Here is my extensions.conf >> [internal] >> exten => 1000,1,Verbose(1|Extension 1000) >> ;exten => 1000,n,Echo() >> ;exten => 1000,n,Hangup() >> exten => 1000,n,Dial(SIP/1000,30) >> exten => 1000,n,Hangup() >> >> exten => 1001,1,Verbose(1|Extension 1001) >> exten => 1001,n,Dial(SIP/1001,30) >> exten => 1001,n,Hangup() >> >> [phones] >> include => internal >> >> >> and sip.conf >> [1000] >> type=friend >> context=phones >> host=dynamic >> [1001] >> type=friend >> context=phones >> host=dynamic >> >> >> I am not setting a password . >> >> Any help will be appreciated. >> >> TIA >> >> Jair Santos >> -- >> >> >> _______________________________________________ >> -- 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 > > ------------------------------ > > > No virus found in this incoming message. > Checked by AVG - www.avg.com > Version: 9.0.698 / Virus Database: 270.14.39/2469 - Release Date: 10/30/09 > 00:52:00 > > > > > _______________________________________________ > -- 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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