Well, for outbound calls, the SIP Server challenges the INVITE with 401/407. Then Re-INVITE is sent which explains why outgoing call works. It is possible that the SIP Server doesnt check to see whether the caller is Registered.
For inbound call, the SIP server needs to know the gateway contact information, and it is obtained only through REGISTER (if not statically configured in the SIP server, which is very unlikely). Thanks, Neel -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean Marc Le Fevre Sent: Thursday, April 19, 2007 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] incoming SIP call Hello and thanks for answering, As I just answer to Yuan LIU, what I don't understand, is that I can place an outbound call from asterisk to a gsm at the same time I can't get asterisk thought a inbound call. But I'll try what you advice me. I'll tell you the result of it Jean-Marc LE FEVRE Le 19 avr. 07 à 20:08, Bala Neelakantan a écrit : If your SIP server loses REGISTERs then it cant place an inbound SIP call. Try changing the REGI STER 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: & lt;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> <-- S IP 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=00-31057-001dc 208-591e1ca81 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] <mailto:[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 quali fy=60000 fromdomain=freephonie.net [freep honie_inbound] type=peer context=incoming host=freephonie.net qualify=60000 allow=all< /P> 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 ; ... _______________________________________________ --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 !DSPAM:4627b30550701698699180! !DSPAM:4627b7bb50703422486060!
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