I'm trying to configure SIP trunking. Now, I'm referencing "Asterisk the definitive guide", 4th ed. While I don't have the page handy, I was reading the suggestion to try SIP to SIP before proceeding to outside connectivity. I'm aware that SIP trunking is a construct, but am, obviously, learning the system.
What I'd like to do is from the CLI "ping" either the peer below, or a peer somewhere. Unfortunately, I'm also in a double+ NAT situation at the moment. While Skype works (mostly) from my LAN, the connection isn't the greatest. My LAN uses a wireless bridge to connect to another LAN. It's just a home setup; it is what it is.
How do I test a connection? How do check the settings? As far as I can tell, the settings are correct.
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 <[email protected]>Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General PublicLicense version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
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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 UNKNOWNbabytel/1<private> 198.38.7.11 D N 5060 UNREACHABLE
gs102/gs102 (Unspecified) D N 0 UNKNOWN3 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
