Faites un appel vers le 06XXX par le peer free, avec le debug.
Faites un appel vers le 06XXX par le peer free, avec le debug.
Je viens de faire le test en passant par le peer free et en activant le debug, il y a absolument rien dans la CLI !Envoyé par Reaper
Le softphone, lui, me dit ceci:
"Vous ne pouvez pas composer ce numéro."
Active le verbose 20
core set verbose 20
Et passe un appel, ensuite regarde le dialplan de contexte "appel_interne_X"
dialplan show appel_interne_X
Vous avez numérote 906XX je suppose ?
Exactement la même chose...
Et oui, j'ai bien utilisé 906XXXXXXXX
SRVVOIP1*CLI> core set verbose 20
Verbosity was 32 and is now 20
SRVVOIP1*CLI>
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '6bd25e715269c1453dc008091a0c2421@DOMAINE' Method: OPTIONS
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90326XXXXXX@appel_interne_X:1] ChanIsAvail("SIP/202-0000000a", "SIP/0326XXXXXX@free-0950XXXXXX") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90326XXXXXX@appel_interne_X:2] Dial("SIP/202-0000000a", "SIP/0326XXXXXX@free-0950XXXXXX),30,r") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [90326XXXXXX@appel_interne_X:3] Congestion("SIP/202-0000000a", "") in new stack
== Spawn extension (appel_interne_X, 90326XXXXXX, 3) exited non-zero on 'SIP/202-0000000a'
SRVVOIP1*CLI>
SRVVOIP1*CLI> dialplan show appel_interne_X
[ Context 'appel_interne_X' created by 'pbx_config' ]
Include => 'salle_de_conference' [pbx_config]
Include => 'local_voicemail' [pbx_config]
Include => 'horloge_parlante' [pbx_config]
Include => 'auto_attendant' [pbx_config]
Include => 'parkedcalls' [pbx_config]
Include => 'client' [pbx_config]
Include => 'globals' [pbx_config]
Include => 'general' [pbx_config]
Include => 'from-asterisk2' [pbx_config]
Include => 'menu' [pbx_config]
Include => 'menu1' [pbx_config]
Include => 'menu2' [pbx_config]
Include => 'menu3' [pbx_config]
Include => 'free-0950XXXXXX' [pbx_config]
Include => 'test' [pbx_config]
-= 0 extensions (0 priorities) in 1 context. =-
SRVVOIP1*CLI>
Pouvez vous retirer les lignes:
exten => _9.,1,ChanIsAvail(SIP/${EXTEN:1}@free-0950XXXXXX)
et la congestion, laisse que le dial. avec priorité 1.
Test ?
Rien du tout !
Je désespères
Appels:
202 -> 90950XXXXXX = La destination non trouvée
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90950XXXXXX@appel_interne_X:1] Dial("SIP/202-00000000", "SIP/0950XXXXXX@free-0950XXXXXX),30,r") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/202-00000000' status is 'CHANUNAVAIL'
SRVVOIP1*CLI>
0950XXXXXX -> 202 = L'indicatif que vous avez demander n'est pas utilisé. Votre appel ne peut aboutir...
Rien dans la CLI
Vous êtes certain que vous avez activé le debug chaque fois sur le peer de free ?
Et autre chose, vous faite un deial vers free-09xxx; il est ou votre peer free-09xxxx ?
exten => _9.,1,Dial(SIP/${EXTEN:1}@0950XXXXXX),30,r)
Oui, en faisant "sip set debug peer 0950XXXXXX"
exten => _9.,1,Dial(SIP/${EXTEN:1}@free-0950XXXXXX),30,r) est dans le context [client]
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90950XXXXXX@appel_interne_X:1] Dial("SIP/202-00000004", "SIP/0950XXXXXX@free-0950XXXXXX),30,r") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/202-00000004' status is 'CHANUNAVAIL'
Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 192.168.X.X:5060;branch=z9hG4bK48d723cf;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@DOMAINE>;tag=as4d70df1b
To: <sip:freephonie.net>
Contact: <sip:Unknown@192.168.X.X>
Call-ID: 5c81be7648e25bcc6a86232355416fd0@DOMAINE
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Date: Mon, 19 Sep 2011 09:13:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
SIP/2.0 501 Not Implemented Yet
Call-ID: 5c81be7648e25bcc6a86232355416fd0@DOMAINE
CSeq: 102 OPTIONS
From: "Unknown" <sip:Unknown@DOMAINE>;tag=as4d70df1b
To: <sip:freephonie.net>;tag=00-30951-1e71b144-6d2ba8477
Via: SIP/2.0/UDP 192.168.X.X:5060;received=82.X.X.X;rport=1024;bran ch=z9hG4bK48d723cf
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '5c81be7648e25bcc6a86232355416fd0@DOMAINE' Method: OPTIONS
SRVVOIP1*CLI>
Encore une fois
cette ligne:
exten => _9.,1,Dial(SIP/${EXTEN:1}@0950XXXXXX),30,r)
Au lieu de:
exten => _9.,1,Dial(SIP/${EXTEN:1}@free-0950XXXXXX),30,r)
Test ?
Ca donne la même chose...(sans le free-)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [90950XXXXXX@appel_interne_X:1] Dial("SIP/202-00000000", "SIP/0950XXXXXX@0950XXXXXX),30,r") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/202-00000000' status is 'CHANUNAVAIL'
Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 192.168.X.X:5060;branch=z9hG4bK64a12463;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@DOMAINE>;tag=as7e3e65bc
To: <sip:freephonie.net>
Contact: <sip:Unknown@192.168.X.X>
Call-ID: 413b10222d56588e1a01d93a255f0608@DOMAINE
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Date: Mon, 19 Sep 2011 09:57:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
SIP/2.0 501 Not Implemented Yet
Call-ID: 413b10222d56588e1a01d93a255f0608@DOMAINE
CSeq: 102 OPTIONS
From: "Unknown" <sip:Unknown@DOMAINE>;tag=as7e3e65bc
To: <sip:freephonie.net>;tag=00-32535-1e73fa81-3df00d4d2
Via: SIP/2.0/UDP 192.168.X.X:5060;received=82.X.X.X;rport=1024;bran ch=z9hG4bK64a12463
Content-Length: 0
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