I'm reading the O'Reilly "Asterisk the definitive guide", 4th ed, with a starfish on it. In some ways, astonishing that it's not really that definitive, it's more general -- and it only clocks in at one ream of paper!
In any event, I'm having some port problems on my home network: http://security.stackexchange.com/questions/81752/ I need to open ports for Asterisk to work even on a local level. so I'm just asking in general. For SIP to SIP peer calling, and by that I just mean "ring" or "beep," some sort of ping, basically, just configure the two softphones to use the IP address for the Asterisk box? also: tleilax:~ # tleilax:~ # asterisk -V Asterisk 1.8.32.1-vici tleilax:~ # tleilax:~ # asterisk -rm Asterisk 1.8.32.1-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others. Created by Mark Spencer <marks...@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= log and verbose output currently muted ('logger mute' to unmute) Connected to Asterisk 1.8.32.1-vici currently running on tleilax (pid = 3062) Verbosity is at least 21 tleilax*CLI> tleilax*CLI> sip show peer babytel * Name : babytel Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Subscr.Cont. : <Not set> Language : en AMA flags : Unknown Netborder CPD: No Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : MOH Suggest : default Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Max forwards : 0 Dynamic : Yes Callerid : "" <> MaxCallBR : 384 kbps Expire : -1 Insecure : no Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: 4294967295 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : Yes TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : sip.babytel.ca Addr->IP : 198.38.7.11:5060 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 1<private> SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw:20) Auto-Framing : No Status : UNREACHABLE Useragent : Reg. Contact : Qualify Freq : 60000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No tleilax*CLI> tleilax*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status 201/201 (Unspecified) D N 0 UNKNOWN babytel/1<private> 198.38.7.11 D N 5060 UNREACHABLE gs102/gs102 (Unspecified) D N 0 UNKNOWN 3 sip peers [Monitored: 0 online, 3 offline Unmonitored: 0 online, 0 offline] tleilax*CLI> thanks, Thufir -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
