Merci pour ton aide,
Voici ce que tu as demandé, j'avoue que c'est un peu long
SIP SHOW PEERS
Code:
telecom*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
maison/maison 93.12.91.104 D N 16654 OK (711 ms)
sfr/014635xxxx 213.91.9.206 N 5060 OK (26 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]
SIP SHOW REGISTRY
Code:
telecom*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
voip.wengo.fr:5060 N kalaza 1785 Registered Mon, 21 Mar 2011 16:23:52
1 SIP registrations.
SIP SET DEBUG IP
Code:
telecom*CLI> sip set debug ip 213.91.9.206
SIP Debugging Enabled for IP: 213.91.9.206
Emission d'appel
Code:
== Using SIP RTP CoS mark 5
-- Executing [061669xxxx@outgoing:1] Dial("SIP/maison-00000000", "SIP/sfr/061669xxxx") in new stack
== Using SIP RTP CoS mark 5
Audio is at MON_IP port 21442
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8000 (slin16) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 213.91.9.206:5060:
INVITE sip:061669xxxx@voip.wengo.fr SIP/2.0
Via: SIP/2.0/UDP MON_IP:5060;branch=z9hG4bK3a4c5572;rport
Max-Forwards: 70
From: "maison" <sip:014635xxxx@MON_IP>;tag=as05e0254f
To: <sip:061669xxxx@voip.wengo.fr>
Contact: <sip:014635xxxx@MON_IP>
Call-ID: 7840143a029de92c23d9af5347a7597d@MON_IP
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.17
Date: Mon, 21 Mar 2011 15:47:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 451
v=0
o=root 1665343704 1665343704 IN IP4 MON_IP
s=Asterisk PBX 1.6.2.17
c=IN IP4 MON_IP
t=0 0
m=audio 21442 RTP/AVP 0 8 3 112 5 10 7 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called sfr/061669xxxx
<--- SIP read from UDP:213.91.9.206:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP MON_IP:5060;branch=z9hG4bK3a4c5572;rport=5060
From: "maison" <sip:014635xxxx@MON_IP>;tag=as05e0254f
To: <sip:061669xxxx@voip.wengo.fr>;tag=b6a81649b79ec740d23d81d957711b9e.090a
Call-ID: 7840143a029de92c23d9af5347a7597d@MON_IP
CSeq: 102 INVITE
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 192.168.70.88:5062 "Noisy feedback tells: pid=7323 req_src_ip=192.168.70.249 req_src_port=5060 in_uri=sip:061669xxxx@voip.wengo.fr out_uri=sip:061669xxxx@voip.wengo.fr via_cnt==2"
<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 213.91.9.206:5060:
ACK sip:061669xxxx@voip.wengo.fr SIP/2.0
Via: SIP/2.0/UDP MON_IP:5060;branch=z9hG4bK3a4c5572;rport
Max-Forwards: 70
From: "maison" <sip:014635xxxx@MON_IP>;tag=as05e0254f
To: <sip:061669xxxx@voip.wengo.fr>;tag=b6a81649b79ec740d23d81d957711b9e.090a
Contact: <sip:014635xxxx@MON_IP>
Call-ID: 7840143a029de92c23d9af5347a7597d@MON_IP
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.17
Content-Length: 0
---
---
-- SIP/sfr-00000001 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/maison-00000000' status is 'CONGESTION'
Really destroying SIP dialog '7840143a029de92c23d9af5347a7597d@MON_IP' Method: INVITE
Reliably Transmitting (NAT) to 213.91.9.206:5060:
OPTIONS sip:voip.wengo.fr SIP/2.0
Via: SIP/2.0/UDP MON_IP:5060;branch=z9hG4bK23038949;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@MON_IP>;tag=as3940a076
To: <sip:voip.wengo.fr>
Contact: <sip:asterisk@MON_IP>
Call-ID: 049ed9754a43d2315b9c0609240d1dde@MON_IP
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.17
Date: Mon, 21 Mar 2011 15:47:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:213.91.9.206:5060 --->
SIP/2.0 405 Method not supported
Via: SIP/2.0/UDP MON_IP:5060;branch=z9hG4bK23038949;rport=5060
From: "asterisk" <sip:asterisk@MON_IP>;tag=as3940a076
To: <sip:voip.wengo.fr>;tag=c563187eab94fcf9eeae89a92b5c0971.c845
Call-ID: 049ed9754a43d2315b9c0609240d1dde@MON_IP
CSeq: 102 OPTIONS
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 192.168.70.249:5060 "Noisy feedback tells: pid=13724 req_src_ip=MON_IP req_src_port=5060 in_uri=sip:voip.wengo.fr out_uri=sip:voip.wengo.fr via_cnt==1"
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '049ed9754a43d2315b9c0609240d1dde@MON_IP' Method: OPTIONS
telecom*CLI>