[asterisk-users] incoming SIP call

[Jean Marc Le Fevre]

Well thanks for answering,

When I test, I use my GSM and call the number my provider gives me.
How often it works or not, I didn't make test like 10 calls per hour  
for a pretty long time so I can't exactly tell. When I test, well  
sometimes it works great, sometime, the incoming call is redirected  
to an phone that is connected on my DSL box.
I didn't see the error message SIP/2.0 403 not registered, but in  
that case:
1) I can make a call from asterisk to a gsm call (so It goes IAX  
phone => asterisk => SIP provider => GSM.
2) if I do show sip register in asterisk CLI, I can see I'm  
registered (or I may be misinterpretting this command.

What can I do to investigate this registration message ? Is there an  
special debug command ?

thanks :)

>> From: Jean Marc Le Fevre <jm.lefevre at etatcritik.dyndns.org>
>> Date: Wed, 18 Apr 2007 18:14:41 +0200
>>
>> 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.
>
> Define "sometimes" and from where the income call you can't get?
>
>> 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
>>
> [good stuff sniffed]
> Where do you suspect the error message is?
>
>> ---
>> Zpro*CLI>
>> <-- SIP read from 212.27.52.5:5060:
>> SIP/2.0 403 not registered
>
> Does this message make sense, "not registered"?
>
> Yuan Liu
>
>> Call-ID: 7263e88c20c9f38c34963cef6704cf07 at 82.XXX.XXX.XXX
>> CSeq: 102 OPTIONS
>> From: "asterisk" <sip:asterisk at 82.XXX.XXX.XXX>;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=z9hG4bK67c2df66
>> Content-Length: 0
>>
>>
>> --- (7 headers 0 lines) ---
>> Destroying call '7263e88c20c9f38c34963cef6704cf07 at 82.XXX.XXX.XXX'
>> Zpro*CLI>
>> <-- SIP read from 212.27.52.5:5060:
>> SIP/2.0 403 not registered
>> Call-ID: 793bf24e5290d562787c8d9451baedd7 at 82.XXX.XXX.XXX
>> CSeq: 102 OPTIONS
>> From: "asterisk" <sip:asterisk at 82.XXX.XXX.XXX>;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=z9hG4bK253c1a3d
>> Content-Length: 0
>>
>> --- (7 headers 0 lines) ---
>> Destroying call '793bf24e5290d562787c8d9451baedd7 at 82.XXX.XXX.XXX'
>>
>>
>> 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
>> allow=ulaw
>> allow=ilbc
>> allow=alaw
>> allow=gsm
>> musicclass=default
>> language=fr
>> useragent=Asterisk PBX
>> dtmfmode = auto
>> register => 09XXXXXXXX:SECRET at freephonie.net
>> 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
>> [freephonie_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
>> ;
>> ...
>>
>>
>>
>>
>>
>>
>>
>>
>>
>
>
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