<div dir="ltr">Dear Sir,<br><br>I have a Patton device registered on an OpenSIPs server...The openSips Server send the call to asterisk server....When using some end user device like patton the asterisk server does not send back an Authentication required packet to INVITE packet received from OS as follow:<br>
<br><--- SIP read from OpenSIPS_IP:5060 ---><br>INVITE sip:0473236354@Asterisk_IP SIP/2.0<br>Record-Route: <sip:OpenSIPS_IP;lr=on;ftag=724c58b04b><br>Via: SIP/2.0/UDP OpenSIPS_IP;branch=z9hG4bK3b3a.4744c247.0<br>
Via: SIP/2.0/UDP 79.132.230.138:5060;rport=5060;received=79.132.230.138;branch=z9hG4bK1f03f72b9bba41567<br>Max-Forwards: 69<br>From: <<a href="http://sip:92968059@79.132.230.138:5060">sip:92968059@79.132.230.138:5060</a>>;tag=724c58b04b<br>
To: <sip:0473236354@OpenSIPS_IP:5060><br>Call-ID: 2ef74e533282d7f1<br>CSeq: 17515 INVITE<br>Contact: <sip:92968059@79.132.230.138:5060;nat=yes><br>Supported: replaces<br>User-Agent: Patton SN4638 5BIS 00A0BA048822 R5.3 2009-01-15 H323 SIP BRI M5T SIP Stack/<a href="http://4.0.28.28">4.0.28.28</a><br>
Content-Type: application/sdp<br>Content-Length: 241<br><br>v=0<br>o=MxSIP 0 2 IN IP4 OpenSIPS_IP<br>s=SIP Call<br>c=IN IP4 OpenSIPS_IP<br>t=0 0<br>m=audio 37586 RTP/AVP 8 0 101<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:0 PCMU/8000<br>
a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=sendrecv<br>a=nortpproxy:yes<br><br><-------------><br>--- (14 headers 12 lines) ---<br>Sending to OpenSIPS_IP : 5060 (no NAT)<br>Using INVITE request as basis request - 2ef74e533282d7f1<br>
[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:650 mysql_reconnect: MySQL RealTime: Everything is fine.<br>[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '92968059'<br>
[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:650 mysql_reconnect: MySQL RealTime: Everything is fine.<br>[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE host = 'OpenSIPS_IP' AND port = '5060'<br>
[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:650 mysql_reconnect: MySQL RealTime: Everything is fine.<br>[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE ipaddr = 'OpenSIPS_IP' AND port = '5060'<br>
[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:650 mysql_reconnect: MySQL RealTime: Everything is fine.<br>[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:258 realtime_multi_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE host = 'OpenSIPS_IP' ORDER BY host<br>
[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:650 mysql_reconnect: MySQL RealTime: Everything is fine.<br>[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:258 realtime_multi_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE ipaddr = 'OpenSIPS_IP' ORDER BY ipaddr<br>
Found no matching peer or user for 'OpenSIPS_IP:5060'<br>Found RTP audio format 8<br>Found RTP audio format 0<br>Found RTP audio format 101<br>[Jul 2 12:44:20] DEBUG[29275]: chan_sip.c:5249 process_sdp: Peer doesn't provide T.38 UDPTL<br>
Peer audio RTP is at port OpenSIPS_IP:37586<br>Found audio description format PCMA for ID 8<br>Found audio description format PCMU for ID 0<br>Found audio description format telephone-event for ID 101<br>Capabilities: us - 0x8080e (gsm|ulaw|alaw|g726|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)<br>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<br>Peer audio RTP is at port OpenSIPS_IP:37586<br>Looking for 0473236354 in BE (domain Asterisk_IP)<br>
[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:650 mysql_reconnect: MySQL RealTime: Everything is fine.<br>[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions WHERE exten = '0473236354' AND context = 'BE' AND priority = '1'<br>
[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:650 mysql_reconnect: MySQL RealTime: Everything is fine.<br>[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:258 realtime_multi_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions WHERE exten LIKE '\\_%' AND context = 'BE' AND priority = '1' ORDER BY exten<br>
[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:650 mysql_reconnect: MySQL RealTime: Everything is fine.<br>[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions WHERE exten = '0473236354' AND context = 'BE-Out' AND priority = '1'<br>
[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:650 mysql_reconnect: MySQL RealTime: Everything is fine.<br>[Jul 2 12:44:20] DEBUG[29275]: res_config_mysql.c:258 realtime_multi_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions WHERE exten LIKE '\\_%' AND context = 'BE-Out' AND priority = '1' ORDER BY exten<br>
list_route: hop: <sip:OpenSIPS_IP;lr=on;ftag=724c58b04b><br><br><--- Transmitting (no NAT) to OpenSIPS_IP:5060 ---><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP OpenSIPS_IP;branch=z9hG4bK3b3a.4744c247.0;received=OpenSIPS_IP<br>
Via: SIP/2.0/UDP 79.132.230.138:5060;rport=5060;received=79.132.230.138;branch=z9hG4bK1f03f72b9bba41567<br>Record-Route: <sip:OpenSIPS_IP;lr=on;ftag=724c58b04b><br>From: <<a href="http://sip:92968059@79.132.230.138:5060">sip:92968059@79.132.230.138:5060</a>>;tag=724c58b04b<br>
To: <sip:0473236354@OpenSIPS_IP:5060><br>Call-ID: 2ef74e533282d7f1<br>CSeq: 17515 INVITE<br>User-Agent: <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Contact: <sip:0473236354@Asterisk_IP><br>
Content-Length: 0<br><br>Please note that if we use linksys as end user SIP device everything looks fine...Even if we register patton on asterisk directly the authentication is successfully done by asterisk<br><br>regards<br>
<br><div class="gmail_quote">On Mon, Jun 29, 2009 at 6:06 PM, Alex Balashov <span dir="ltr"><<a href="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
It does this by default unless you have allowguest set to yes, and/or<br>
any insecure parameter options on any individual peers.<br>
<br>
--<br>
Sent from mobile device<br>
<div><div></div><div class="h5"><br>
On Jun 29, 2009, at 10:33 AM, michel freiha <<a href="mailto:michofr@gmail.com">michofr@gmail.com</a>> wrote:<br>
<br>
> Hi all,<br>
><br>
> i would like to ask please about how to force asterisk to ask for<br>
> authentication when receiving an INVITE packet from any device?<br>
><br>
> Regards<br>
</div></div>> _______________________________________________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
><br>
> asterisk-users mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>
> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
<br>
_______________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</blockquote></div><br></div>