bonjour, je suis passé a la version 8, mais toujours le même problème (pas de son) :'(, et meme un nouveau, car le client EKIGA à un message "le controle de sécurité a échoué".
Code:
Name/username Host Dyn Forcerport ACL Port Status
HClO/HClO 192.168.1.10 D 5060 UNREACHABLE
SPA/SPA 192.168.1.253 D 5060 Unmonitored
SPAFIXE (Unspecified) D 0 Unmonitored
fixevoip/fixevoip 82.124.42.*** D N 1024 OK (173 ms)
4 sip peers [Monitored: 1 online, 1 offline Unmonitored: 1 online, 1 offline]
une difference dans les IP entre HClO et fixevoip, alors qu' il sont dans le meme réseau .
(note il y a une différence : HClO est en nat=comedia)
Code:
jn-serveur*CLI> sip show peer HClO
* Name : HClO
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : local
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : Yes
Callerid : "HClO" <200>
MaxCallBR : 384 kbps
Expire : 3162
Insecure : no
Force rport : No
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 192.168.1.10:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: HClO
SIP Options : (none)
Codecs : 0x80c (ulaw|alaw|g726)
Codec Order : (ulaw:20,alaw:20,g726:20)
Auto-Framing : No
100 on REG : No
Status : UNREACHABLE
Useragent : Ekiga/3.2.7
Reg. Contact : sip:HClO@192.168.1.10
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
Code:
jn-serveur*CLI> sip show peer fixevoip
* Name : fixevoip
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : local
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : Yes
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : 100
Insecure : no
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 82.124.42.***:1024
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: fixevoip
SIP Options : (none)
Codecs : 0x80c (ulaw|alaw|g726)
Codec Order : (ulaw:20,alaw:20,g726:20)
Auto-Framing : No
100 on REG : No
Status : OK (175 ms)
Useragent : C470 IP/022270000000
Reg. Contact : sip:fixevoip@192.168.1.12:5060
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
Code:
jn-serveur*CLI> sip show settings
Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: No
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: Asterisk PBX 1.8.6.0
SDP Session Name: Asterisk PBX 1.8.6.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Legacy userfield parse: No
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Network QoS Settings:
---------------------------
IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Network Settings:
---------------------------
SIP address remapping: Enabled using externaddr
Externhost: <none>
Externaddr: 78.226.117.**:5060
Externrefresh: 10
Localnet: 192.168.1.0/255.255.255.0
Global Signalling Settings:
---------------------------
Codecs: 0x80c (ulaw|alaw|g726)
Codec Order: ulaw:20,alaw:20,g726:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: default
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
----
pour ma part je me demande pourquoi les Reg. Contact on l' IP local
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