[asterisk-users] incoming SIP call

Thu Apr 19 11:40:55 MST 2007

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 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: asterisk-users-bounces at lists.digium.com [mailto:asterisk- 
> users-bounces at lists.digium.com] On Behalf Of Jean Marc Le Fevre
> Sent: Wednesday, April 18, 2007 11:15 AM
> To: asterisk-users at lists.digium.com
> 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:asterisk at 82.XXX.XXX.XXX>;tag=as01265eaf
>
> To: <sip:freephonie.net>
>
> Contact: <sip:asterisk at 82.XXX.XXX.XXX>
>
> Call-ID: 7263e88c20c9f38c34963cef6704cf07 at 82.XXX.XXX.XXX
>
> 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:asterisk at 82.XXX.XXX.XXX>;tag=as372da2cb
>
> To: <sip:freephonie.net>
>
> Contact: <sip:asterisk at 82.XXX.XXX.XXX>
>
> Call-ID: 793bf24e5290d562787c8d9451baedd7 at 82.XXX.XXX.XXX
>
> 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 8c34963cef6704cf07 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: "asteris k" <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
>
> all ow=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
>
> [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
>
> ;
>
> ...
>
>
>
>
>
>
>
>
>
> 
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