Question

我无法使音频在FortiGate防火墙以外的星号服务器中使用sip配置。

电话正传给zoiper，但没有音频。

星号版本：16.5.1

以下是我的sip配置：

[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=udp,tcp
srvlookup=yes
alwaysauthreject=yes
allowguest=no
qualify=yes
bindport=5080
canreinvite=no

externip=182.74.18.xxx
localnet=192.168.10.0/255.255.255.0
nat=force_rport,comedia
directmedia=no
insecure=port,invite

defaultexpiry=600
progressinband=yes

[client](!)
type=friend
disallow=all
allow=gsm
allow=ulaw
allow=alaw
dtmfmode=rfc2833
sendrpid=yes
trustrpid=yes
qualify=yes
directmedia=no
directrtpsetup=no
canreinvite=no
host=dynamic
insecure=port,invite

[2503](client)
context=intercom
secret=xxxxx

对等详细信息

asterisk*CLI> sip show peer 2503


  * Name       : 2503
  Description  : 
  Realtime peer: No
  Secret       : <Not set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : intercom
  Record On feature : automon
  Record Off feature : automon
  Subscr.Cont. : <Not set>
  Language     : 
  Tonezone     : <Not set>
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  Named Callgr : 
  Nam. Pickupgr: 
  MOH Suggest  : 
  Mailbox      : 
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Max forwards : 0
  Dynamic      : Yes
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : port,invite
  Force rport  : Yes
  Symmetric RTP: Yes
  ACL          : No
  ContactACL   : No
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: 4294967295
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : Yes
  Send RPID    : Yes
  Path support : No
  Path         : N/A
  TrustIDOutbnd: Legacy
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : 
  Addr->IP     : (null)
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP,TCP
  Def. Username: 2503
  SIP Options  : (none)
  Codecs       : (gsm|ulaw|alaw)
  Auto-Framing : No
  Status       : UNKNOWN
  Useragent    : Z 5.2.28 rv2.8.115
  Reg. Contact : sip:2503@192.168.10.1:42820;transport=UDP;rinstance=f47bdfb0c923d335
  Qualify Freq : 60000 ms
  Keepalive    : 0 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   : 
  Use Reason   : No
  Encryption   : No
  RTCP Mux     : No

我尝试使用以下设置更改防火墙规则。但没有运气

config system session-helper
delete 13
delete 3
delete 2
end

- disable sip helper and nat trace
config system settings
set sip-helper disable
set sip-nat-trace disable
end

- edit voip profile
config voip profile
edit default config
sip set status disable
end
end 

- Flush ARP cache
execute clear system arp table

- nuclear option, reset all sessions
diagnose sys session clear

过去2天我无法使其正常工作。

Asterisk SIP没有超出增强的60E防火墙的音频

0 个答案: