If your SIP server loses REGISTERs then it cant place an inbound SIP call. Try changing the REGISTER frequency to lower value.
When you see incoming SIP call fail, you might want to check whether the REGISTERs are working. Thanks, Neel -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean Marc Le Fevre Sent: Wednesday, April 18, 2007 11:15 AM To: [email protected] Subject: [asterisk-users] incoming SIP call Hello all, I'm having a quite simple configuration like: SIP provider <=> asterisk SIP <=> lan Everythings works fine but sometime I can't get incoming call. here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch= z9hG4bK67c2df66;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as01265eaf To: <sip:freephonie.net> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 12 headers, 0 lines Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;b ranch=z9hG4bK253c1a3d;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as372da2cb To: <sip:freephonie.net> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 18 Apr 2007 13:57:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Zpro*CLI> <-- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: 7263e88c20c9f3 <mailto:[EMAIL PROTECTED]> [EMAIL PROTECTED] CSeq: 102 OPTIONS From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as01265eaf To: <sip:freephonie.net>;tag=00-31057-001dc208-591e1ca81 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df6 6 Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Zpro*CLI> <-- SIP read from 212.27.52.5:5060: SIP/2.0 403 not registered Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS From: "asteris k" <sip:[EMAIL PROTECTED]>;tag=as372da2cb To: <sip:freephonie.net>;tag=00-32700-001dc209-6fc2b3303 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3 d Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' sip.conf [general] context=incoming realm=etatcritik.dyndns.org bindport=5060 bindaddr=0.0.0.0 srvlookup=no maxexpiry=3600 defaultexpiry=1800 videosupport=yes disallow=all all ow=ulaw allow=ilbc allow=alaw allow=gsm musicclass=default language=fr useragent=Asterisk PBX dtmfmode = auto register => 09XXXXXXXX:[EMAIL PROTECTED] registertimeout=40 externip = 82.XXX.XXX.XXX localnet=10.XXX.XXX.XXX/255.255.255.0 qualify=60000 nat = yes [test] type=friend username=test secret=test host=dynamic context=home callerid =test <2222> dmtfmode=rfc2833 authuser=test fromuser=test allow=all [freephonie_outbound] type=peer allow=all host=freephonie.net secret=SECRET fromuser=09XXXXXXX username=09XXXXXXX dtmfmode=inband qualify=60000 fromdomain=freephonie.net [freep honie_inbound] type=peer context=incoming host=freephonie.net qualify=60000 allow=all deny=0.0.0.0/0..0.0.0 permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net etension.conf ... [incoming] exten => s,1,Ringing exten => s,2,Noop(I receive a sip call); exten => s,n,Goto(home,1000,1) exten => s,n,Congestion ; ... !DSPAM:462643f450705772331342!
_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
