【发布时间】：2014-05-22 12:15:43
【问题描述】：
我正在尝试在星号和 Skype 连接之间进行通信。我有一个注册了 1 个频道的 sip 个人资料。此外，星号注册到 sip.skype.com 好的。从 CLI：
sip 显示同行
skype/99051000XXXXXX 63.209.144.201 Yes Yes 5060 OK (178 ms)
sip 显示注册表
Reg.Time
sip.skype.com:5060 N 99051000XXXX 30 Registered
Tue, 08 Apr 2014 22:18:29
1 SIP registrations.
但是，每当我尝试拨打 Skype 电话时，我都会收到：
*CLI> == Using SIP RTP CoS mark 5
-- Executing [4321@LocalSets:1] Dial("SIP/ZOI6001-00000038", "SIP/skype/4321") in new
stack
== Using SIP RTP CoS mark 5
-- Called SIP/skype/4321
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/ZOI6001-00000038' status is 'CHANUNAVAIL'
出于某种原因，我想我应该提一下，每当我重新加载 sip 时，它会说
`Using Cos mark 4`
而不是像上面那样的 Cos 标记 5。
在 sip.conf 中：
[general]
;register => gnext.telephony:xxxxxxxxCfnum@sip.skype.com/99051000234871
context=from-trunk
allowoverlap=no
;allowguest=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
allow=all
dtmfmode=rfc2833
;bindport = 56782
port=5060
;register => gnext.telephony:xxxxxxxxxnum@sip.skype.com/99051000234871
register =>99051000234871:xxxxxxxxxxx@sip.skype.com/99051000234871
;register => 9905100xxxxxxx:xxxxxxxxxx@sip.skype.com/9905100xxxxxxxxx
trustrpid=no
sendrpid=yes
calllimit=4
defaultexpiry=240
[skype]
type=friend
;type=peer
;context=from-trunk
context=from-trunk
username=9905xxxxxxx
secret=xxxxxxxx
canreinvite=no
insecure=port,invite
dtmfmode=rfc2833
host=sip.skype.com
nat=force_rport,comedia
qualify=yes
fromuser=xxxxxxxxxx
fromdomain=sip.skype.com
disallow=all
allow=g729
allow=ulaw
allow=alaw
我的拨号方案：
[from-trunk]
exten => 1234,n,Dial(SIP/XLX6003,15);
[from-local]
exten => 4321,1,Dial(SIP/skype/${EXTEN})
sip debug (sip set debug on) 显示：
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '4e3bd3734b8db5a35bf963ee50b0e2b3@192.168.0.103:5060'
Method: OPTIONS
[Apr 8 22:26:18] NOTICE[12076]: chan_sip.c:15059 sip_reregister:-- Re-registration
for 99051000234871@sip.skype.com
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 63.209.144.201:5060:
REGISTER sip:sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK06df1bdd
Max-Forwards: 70
From: <sip:99051000234871@sip.skype.com>;tag=as22c0d1e3
To: <sip:99051000234871@sip.skype.com>
Call-ID: 53a73ba5083511cb29b1d3cf6bf4f37b@[::1]
CSeq: 109 注册 用户代理：Asterisk PBX 11.8.1 过期：120 接触： 内容长度：
有什么想法吗？请告诉我。 谢谢！！
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