???
???
=> je vois pas le pbm dans le traces, le ignoring est plutot du à une retransmission, est ce systematique ?
=> le /NUM_TEL
il faut je pense qu'il y ait une extension [NUM_TEL] dans le dialplan pour que cela marche...
Code:; Asterisk can register as a SIP user agent to a SIP proxy (provider) ; Format for the register statement is: ; register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] ;
plus précisemment, dans le contexte de reception des appels, tu dois avoir l'extension /NUM_TEL
exten => NUM_TEL,....
j'ai mit 336********
quand j'appelle:
Mode debug:Code:Asterisk 1.6.2.5-0ubuntu1.3, Copyright (C) 1999 - 2009 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.5-0ubuntu1.3 currently running on Sediad (pid = 741) Verbosity was 0 and is now 7 == Using SIP RTP CoS mark 5 Sediad*CLI>
Code:SIP Debugging enabled Sediad*CLI> <--- SIP read from UDP:213.215.45.230:5060 ---> INVITE sip:331********@***.***.***.*** SIP/2.0 Record-Route: <sip:213.215.45.230;lr=on> Via: SIP/2.0/UDP 213.215.45.230;branch=z9hG4bK4021.e4e3eba1.0 Via: SIP/2.0/UDP 213.215.45.252:5060;received=213.215.45.252;branch=z9hG4bK4e499720;rport=5060 From: "331********" <sip:331********@ippi.fr>;tag=as259f1013 To: <sip:01********@ippi.fr> Contact: <sip:331********@213.215.45.252> Call-ID: 535a09dd7b98f0d231326e4613d941ab@ippi.fr CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 12 Date: Fri, 01 Jul 2011 08:11:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 340 DID-info: 331******** v=0 o=root 17453 17453 IN IP4 213.215.45.252 s=session c=IN IP4 213.215.45.252 t=0 0 m=audio 19328 RTP/AVP 8 0 97 3 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> --- (16 headers 15 lines) --- == Using SIP RTP CoS mark 5 Sending to 213.215.45.230 : 5060 (no NAT) Using INVITE request as basis request - 535a09dd7b98f0d231326e4613d941ab@ippi.fr Found peer 'ippi' for '331********' from 213.215.45.230:5060 <--- Reliably Transmitting (NAT) to 213.215.45.230:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 213.215.45.230;branch=z9hG4bK4021.e4e3eba1.0;received=213.215.45.230 Via: SIP/2.0/UDP 213.215.45.252:5060;received=213.215.45.252;branch=z9hG4bK4e499720;rport=5060 From: "331********" <sip:331********@ippi.fr>;tag=as259f1013 To: <sip:01********@ippi.fr>;tag=as26827315 Call-ID: 535a09dd7b98f0d231326e4613d941ab@ippi.fr CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.5-0ubuntu1.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e0ad280" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '535a09dd7b98f0d231326e4613d941ab@ippi.fr' in 6400 ms (Method: INVITE) Sediad*CLI> <--- SIP read from UDP:213.215.45.230:5060 ---> ACK sip:331********@***.***.***.*** SIP/2.0 Via: SIP/2.0/UDP 213.215.45.230;branch=z9hG4bK4021.e4e3eba1.0 From: "331********" <sip:331********@ippi.fr>;tag=as259f1013 Call-ID: 535a09dd7b98f0d231326e4613d941ab@ippi.fr To: <sip:01********@ippi.fr>;tag=as26827315 CSeq: 102 ACK Max-Forwards: 70 User-Agent: OpenSIPS (1.6.3-notls (i386/linux)) Content-Length: 0 <-------------> --- (9 headers 0 lines) ---
Je remet mon code actuelle:
sip.conf:
extensions.conf;Code:[general] defaultexpirey=1800 dtmfmode=auto qualify=yes register => UTILISATEUR:MOT_DE_PASSE@ippi.fr/NUM_TEL ;NUM_TEL ecrit 331******** externip=sediad.homelinux.org externrefresh=60 localnet=192.168.1.0/255.255.255.0 [ippi] type=friend host=ippi.fr username=UTILISATEUR secret=MOT_DE_PASSE fromuser=UTILISTATEUR fromdomain=ippi.fr nat=yes canreinvite=no insecure=very context=fromippi [sediad1] type=friend secret=MOT_DE_PASSE host=dynamic context=home nat=yes [sediad2] type=friend secret=MOT_DE_PASSE host=dynamic context=home nat=yes
Code:[fromippi] exten => _+.,1,Verbose(1, *** International Formati : 00${EXTEN:1}) exten => s,n,Dial(SIP/sediad1) [home] exten => 001,1,Dial(SIP/sediad1) exten => 002,1,Dial(SIP/sediad2) exten => _X.,1,Dial(SIP/ippi/${EXTEN})
Bonjour,
Voici comment j'ai configuré chez moi pour ippi.
Code:[ippi_incoming] ; configuration des appels entrants depuis ippi type=peer host=ippi.fr context=contextVoilà chez moi tout fonctionne.Code:[ippi_appel-sortant-ippi] ; configuration des appels sortants par ippi type=peer host=ippi.fr username=username secret=mdp fromuser=user fromdomain=ippi.fr nat=yes insecure=port,invite
Cordialement Vincent.
Version asterisk: Asterisk 1.6.2.9
Merci vincent10
j'ai avancé d'un pas, je ne pence pas être loin de la vérité:
Lorsque j'appelle j'ai l'erreur suivante:
Donc, j'ai donc retirer l’extension dans "register", sa me donne donc:Code:== Using SIP RTP CoS mark 5 [Jul 2 20:25:47] NOTICE[873]: chan_sip.c:20039 handle_request_invite: Call from 'ippi_incoming' to extension '331********' rejected because extension not found.
j'ai donc l'erreur suivante:Code:register => UTILISATEUR:MOT_DE_PASSE@ippi.fr
Code:== Using SIP RTP CoS mark 5 [Jul 3 21:18:59] NOTICE[995]: chan_sip.c:20039 handle_request_invite: Call from 'ippi_incoming' to extension 's' rejected because extension not found.
Sa me donne :
sip.conf :Code:== Using SIP RTP CoS mark 5 [Jul 4 09:45:56] NOTICE[892]: chan_sip.c:20039 handle_request_invite: Call from 'ippi_incoming' to extension '01********' rejected because extension not found.
extensions.conf :Code:[general] defaultexpirey=1800 dtmfmode=auto qualify=yes register => UTILISATEUR:MOT_DE_PASSE@ippi.fr/01******** externip=sediad.homelinux.org externrefresh=60 localnet=192.168.1.0/255.255.255.0 [ippi_incoming] ; configuration des appels entrants depuis ippi type=peer host=ippi.fr context=context [ippi_appel-sortant-ippi] ; configuration des appels sortants par ippi type=peer host=ippi.fr username=UTILISATEUR secret=MOT_DE_PASSE fromuser=UTILISATEUR fromdomain=ippi.fr nat=yes insecure=port,invite [sediad1] type=friend secret=mot_de_passe host=dynamic context=home nat=yes [sediad2] type=friend secret=mot_de_passe host=dynamic context=home nat=yes
Code:[fromippi] exten => 01********,1,Ringing(1) exten => 01********,n,Wait(3) exten => 01********,n,Answer exten => 01********,n,Dial(SIP/sediad1) [home] exten => 001,1,Dial(SIP/sediad1) exten => 002,1,Dial(SIP/sediad2) exten => _00.,1,Dial(SIP/ippi_appel-sortant-ippi/${EXTEN})
Dans [ippi_incoming]
le context est bien fromippi?
Cordialement Vincent.
Version asterisk: Asterisk 1.6.2.9