返回列表 發帖
之前Qnewbie 兄說在Asterisk PSTN 方面係work 的,但在SIP 方面就唔work,其實你PSTN 定 SIP 問題呢?

IP01在Silence Detection 方面其實沒有太多調較空間,反而在SPA3102 還有些地方可以控制一下。

TOP

你超高 line-voltage 的問題解決咗未?  呢樣野都影響SIP接出接入掛線問題的。

TOP

本帖最後由 bubblestar 於 2011-3-5 17:57 編輯

IP01 SIP.conf 基本上我跟你的差不多,但rtpholdtime out 一定要大過 rtptime out 的,以前我set 120/60,現在用 300/60。

至於你之前超高 SPA3102 line voltage,老實說一定有影響。 而且聽你說現在是借公司線用,公司又是用緊PBX,情況較複雜,很難推想問題出在那裡。
我現在沒有插SPA3102使用,稍後再拿出來看看SETTINGS,到時再試下睇下是否幫到你手。


[general]
context = default
allowoverlap = no
bindport = I use a port number other than 5060 in my case
bindaddr = 0.0.0.0
srvlookup = yes
allowexternaldomains = yes
allowguest = no
allowsubscribe = yes
allowtransfer = yes
alwaysauthreject = yes
autodomain = no
callevents = no
canreinvite =
checkmwi = 10
compactheaders = no
defaultexpiry = 120
domain =
dtmfmode = auto
dumphistory = no
externrefresh = 10
fromdomain = This is set to be my DDNS address
g726nonstandard = no
jbenable = no
jbforce = no
jbimpl =
jblog = no
jbmaxsize =
jbresyncthreshold =
language = en
maxcallbitrate = 384
maxexpiry = 3600
minexpiry = 60
mohinterpret = default
mohsuggest =
nat = yes
notifyringing = yes
pedantic = no
progressinband = never
promiscredir = no
realm = asterisk
recordhistory = no
registerattempts = 0
registertimeout = 20
relaxdtmf = yes
rtpholdtimeout = 300
rtptimeout = 60

sendrpid = no
sipdebug = no
subscribecontext =
t1min = 100
t38pt_udptl = yes
tos_audio = ef
tos_sip = CS3
tos_video = AF41
trustrpid = no
useragent = SwitchFin PBX
usereqphone = no
videosupport = yes
externhost = This is also set to my DDNS address
localnet = my local net address/255.255.255.0
disallow = all
allow = alaw,ulaw,gsm

TOP

你localnet 一定要SET netmask,而且不是SET 梗咗一個 local IP address 的。

localnet = 192.168.0.38 (wrong setting)

localnet = 192.168.0.0/255.255.255.0 (whole lan segment with netmask is appropriate)

怪不得你經常SIP 的網絡有問題。

TOP

如果設定無誤,正常應該唔需要externhost 及 externip 一齊用的。

TOP

Yes, correct.

TOP

回復 27# smallstarxsc


You are welcome.

Happy sharing

TOP

返回列表