Kamailio (192.168.182.1) and Asterisk (192.168.182.24) realtime integration by http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb.

Kamailio have a public interface also.

Grep from debug on kamailio 4.2.1 on call from asterisk extensions/peers:

Mar 19 14:51:50 netbox /usr/sbin/kamailio[547784]: DEBUG: usrloc [udomain.c:602]: db_load_urecord(): aor 101@192.168.182.1 not found in table location Mar 19 14:51:50 netbox /usr/sbin/kamailio[547784]: DEBUG: registrar [lookup.c:158]: lookup(): '101@192.168.182.1' Not found in usrloc

Mar 19 14:51:50 netbox /usr/sbin/kamailio[547785]: DEBUG: usrloc [udomain.c:602]: db_load_urecord(): aor 102@192.168.182.1 not found in table location Mar 19 14:51:50 netbox /usr/sbin/kamailio[547785]: DEBUG: registrar [lookup.c:158]: lookup(): '102@192.168.182.1' Not found in usrloc

Kamailio forwards registration to Asterisk and store location of peer with domain name sip.domain.tld. But Asterisk dials peer to peer in local dialplan - to IP of kamailio.

asterisk1*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 101/101 192.168.182.1 D No No A 5060 Unmonitored Cached RT 102/102 192.168.182.1 D No No A 5060 Unmonitored Cached RT

asterisk1*CLI> sip show peer 102

* Name : 102 Description : Realtime peer: Yes, cached Secret : <Not set> MD5Secret : <Not set> Remote Secret: <Not set> Context : ext-local Record On feature : automon Record Off feature : automon Subscr.Cont. : <Not set> Language : Tonezone : <Not set> AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened FromUser : 102 FromDomain : sip.domain.com Port 5060 Callgroup : Pickupgroup : Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 6/0 Call limit : 0 Max forwards : 0 Dynamic : Yes Callerid : "" <> MaxCallBR : 384 kbps Expire : 252 Insecure : no Force rport : No Symmetric RTP: No ACL : Yes DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : No Path support : No Path : N/A Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 192.168.182.1:5060 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 102 SIP Options : (none) Codecs : (gsm|ulaw|alaw|h263|testlaw) Codec Order : (none) Auto-Framing : No Status : Unmonitored Useragent : kamailio (4.1.2 (x86_64/linux)) Reg. Contact : sip:102@192.168.182.1:5060 Qualify Freq : 60000 ms Keepalive : 0 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No

/etc/asterisk/extensions.conf:

... [ext-local] ; our phones use 3 digit extensions, starting with 1 exten => _1XX,1,DumpChan(verbose) exten => _1XX,n,Dial(SIP/${EXTEN}) exten => _1XX,n,Voicemail(${EXTEN},u) exten => _1XX,n,Hangup exten => _1XX,101,Voicemail(${EXTEN},b) exten => _1XX,102,Hangup

Also strabge why I see user.pub.lic in Variables on channel during call: ... SIPCALLID=GOB1.VSLU0SOTjG-u8jjarzXv.lJTjin SIPDOMAIN=netbox.teleservice.com.ua SIPURI=sip:101@user.pub.lic.ip:35835

Do I need rewrite To header with change IP to domain name? Maybe in this section:

#!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; }