It looks as if that is more of a question/issue with your router, rather than Asterisk.
I have SIP devices working on my LAN, all hardwired, and have no need to open any ports or have the router address SIP in any way My switch is not managed, and the router ports on the LAN side are all unmanaged, just a huge Ethernet "wirenut" You SHOULD be able to communicate between devices on the LAN without any firewall issue. I have also found with some routers that the DMZ isn't what one expects, and can get in the way, depending on the firware. Does this router have any SIP ALG setting? turn it off! As an aside, I would caution you to not have SIP 5060 exposed to the public Internet, or you will soon regret it. I am sure others will have much better information though John Novack thufir wrote:
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
-- Dog is my Co-pilot
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