All,<br><br>I have just setup kamailio as SPI outbound proxy, in front of Asterisk.<br><br>I'm novice with Kamailio, it's the first time I use it.<br><br>The setup is working but I need your advises:<br><br>1) When I type the "sip show peers" command in ASterisk, I see the ip address of the sip proxy. The qualify (monitoring/keepalive) seems to be
sent to the sip proxy and not to the phone. Is there an alternative to
directly monitor the phone ?<br>
<br>localhost*CLI> sip show peers<br>phone2a/phone2a <b> <IP_SIP_PROXY> </b> D N 53 OK (299 ms) <div id=":2gs"> <br><br><br>2) If I use the command "localhost*CLI> sip show peer phone2a"<br>
The ip of the phone is visible in the field "Reg. Contact" only.<br>In the field " Addr->IP", it's the IP of the SIP Proxy.<br><br> * Name : phone2a<br>
Secret : <Set><br> MD5Secret : <Not set><br> Remote Secret: <Not set><br> Context : client2<br> Subscr.Cont. : <Not set><br> Language :<br> AMA flags : Unknown<br> Transfer mode: open<br>
CallingPres : Presentation Allowed, Not Screened<br> Callgroup :<br> Pickupgroup :<br> MOH Suggest :<br> Mailbox :<br> VM Extension : *97<br> LastMsgsSent : 32767/65535<br> Call limit : 0<br> Max forwards : 0<br>
Dynamic : Yes<br> Callerid : "" <><br> MaxCallBR : 384 kbps<br> Expire : 554<br> Insecure : no<br> Force rport : Yes<br> ACL : No<br> DirectMedACL : No<br> T.38 support : No<br>
T.38 EC mode : Unknown<br> T.38 MaxDtgrm: -1<br> DirectMedia : No<br> PromiscRedir : No<br> User=Phone : No<br> Video Support: No<br> Text Support : No<br> Ign SDP ver : No<br> Trust RPID : No<br> Send RPID : No<br>
Subscriptions: Yes<br> Overlap dial : Yes<br> DTMFmode : rfc2833<br> Timer T1 : 500<br> Timer B : 32000<br> ToHost :<br> Addr->IP : <b><SIP PROXY IP></b><br>
Defaddr->IP : (null)<br> Prim.Transp. : UDP<br> Allowed.Trsp : UDP<br> Def. Username: phone2a<br> SIP Options : (none)<br> Codecs : 0xa (gsm|alaw)<br> Codec Order : (gsm:20,alaw:20)<br> Auto-Framing : No<br>
Status : OK (299 ms)<br> Useragent : LinphoneAndroid/2.1.2 (eXosip2/3.6.0)<br> Reg. Contact : sip:phone2a@<b><IP PHONE></b><br><div id=":2gs">line=8b1b24fbaaf794a<br> Qualify Freq : 60000 ms<br> Sess-Timers : Accept<br>
Sess-Refresh : uas<br>
Sess-Expires : 1800 secs<br> Min-Sess : 90 secs<br> RTP Engine : asterisk<br> Parkinglot :<br> Use Reason : No<br> Encryption : No</div><br>3) I'm running a fail2ban protection to protect
against scanners and my fail2ban is blocking the SIP Proxy when the threshold is reached, which means that all the clients behind the sip
outbound proxy are blocked.<br>I think the points 1, 2, 3 are related and if the SIP Proxy could be "transparent"<br><br><br>Here is a debug of a register request, taken on the kamailio server<br>REGISTER sip:<a href="http://pbx-qa.mydomain.com">pbx-qa.mydomain.com</a> SIP/2.0<br>
Via: SIP/2.0/UDP <IP SIP PROXY>;branch=z9hG4bKfea4.34a8fd83.0<br>Via: SIP/2.0/UDP 100.96.196.103:4294;received=<IP PHONE>;rport=6738;branch=z9hG4bK1329841729<br>From: <<a href="mailto:sip%3Aphone2a@mydomain.com">sip:phone2a@mydomain.com</a>>;tag=1870152222<br>
To: <<a href="mailto:sip%3Aphone2a@pbx-qa.domain.com">sip:phone2a@pbx-qa.domain.com</a>><br>Call-ID: 1455540209<br>CSeq: 2 REGISTER<br>Contact: <sip:phone2a@100.96.196.103:4294;line=c6f956d7cdb0eb5><br>Authorization: Digest username="phone2a", realm="asterisk", nonce="176e8fa1", uri="sip:<a href="http://pbx-qa.domain.com">pbx-qa.domain.com</a>", response="4c68e98p1ea7cb0ee81674a8384ca6e4", algorithm=MD5<br>
Max-Forwards: 69<br>User-Agent: LinphoneAndroid/2.1.2 (eXosip2/3.6.0)<br>Expires: 600<br>Content-Length: 0<br>P-hint: outbound<br><br><br><br>Any ideas to solve my problem, to get a "more" transparent proxy ?<br>
<br>Regards,<br>Renaud Dubois<br></div>