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发帖数: 8453 | 1 参考前面帖子在pogo上跑了asterisk服务。用IPTel和IPKall的免费服务和号码。前几
天还好，昨天突然dumb平板上的软电话频繁收到google voice转的电话(206xxxxxx)。
按接听也听不见声音。看GV历史记录发现全是一些陌生号码，基本间隔一两分钟一个。
有的还是收费电话，GV扣了钱。奇怪的是除了软电话GV连的其他电话都不响。停了
asterisk server之后软电话就不提示有电话进来了。打开
asterisk console看见一堆转接信息，也不知道啥意思。下面贴几个例子：
== Parsing '/etc/asterisk/asterisk.conf': == Found
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
== Parsing '/etc/asterisk/extconfig.conf': == Found
Connected to Asterisk 1.6.2.9-2+squeeze12 currently running on debian (pid =
3239)
Verbosity is at least 3
== Using SIP RTP CoS mark 5
-- Executing [12462562708@phone:1] Answer("SIP/192.168.2.113:5060-
0000008e", "") in new stack
-- Executing [12462562708@phone:2] Set("SIP/192.168.2.113:5060-0000008e"
, "GLOBAL(gvuser)=SIP/192.168.2.113:5060-0000008e") in new stack
== Setting global variable 'gvuser' to 'SIP/192.168.2.113:5060-0000008e'
-- Executing [12462562708@phone:3] System("SIP/192.168.2.113:5060-
0000008e", "gvoice -e z**[email protected] -p xxxxxx call 12462562708 12538000000
1 &") in new stack
-- Executing [12462562708@phone:4] Ringing("SIP/192.168.2.113:5060-
0000008e", "") in new stack
-- Executing [12462562708@phone:5] Wait("SIP/192.168.2.113:5060-0000008e
", "30") in new stack
[May 25 00:13:54] WARNING[3252]: chan_sip.c:3915 retrans_pkt: Maximum
retries exceeded on transmission 8f5ba5421923a41e for seqno 1 (Critical
Response) -- See doc/sip-retransmit.txt.
[May 25 00:13:54] WARNING[3252]: chan_sip.c:3942 retrans_pkt: Hanging up
call 8f5ba5421923a41e - no reply to our critical packet (see doc/sip-
retransmit.txt).
== Spawn extension (phone, 12687803282, 5) exited non-zero on 'SIP/192.168
.2.113:5060-0000008c'
-- Executing [h@phone:1] GotoIf("SIP/192.168.2.113:5060-0000008c", "0?:
bridged") in new stack
-- Goto (phone,h,6)
-- Executing [h@phone:6] NoOp("SIP/192.168.2.113:5060-0000008c", "The
channel has been bridged successfully") in new stack
-- Executing [h@phone:7] Set("SIP/192.168.2.113:5060-0000008c", "GLOBAL(
gvuser)="zzzh"") in new stack
== Setting global variable 'gvuser' to '"zzzh"'
== Using SIP RTP CoS mark 5
-- Executing [12462550260@phone:1] Answer("SIP/192.168.2.113:5060-
0000008f", "") in new stack
-- Executing [12462550260@phone:2] Set("SIP/192.168.2.113:5060-0000008f"
, "GLOBAL(gvuser)=SIP/192.168.2.113:5060-0000008f") in new stack
== Setting global variable 'gvuser' to 'SIP/192.168.2.113:5060-0000008f'
-- Executing [12462550260@phone:3] System("SIP/192.168.2.113:5060-
0000008f", "gvoice -e z**[email protected] -p xxxxxx call 12462550260 12538000000
1 &") in new stack
-- Executing [12462550260@phone:4] Ringing("SIP/192.168.2.113:5060-
0000008f", "") in new stack
-- Executing [12462550260@phone:5] Wait("SIP/192.168.2.113:5060-0000008f
", "30") in new stack
-- Executing [12645828167@phone:6] NoOp("SIP/192.168.2.113:5060-0000008d
", "Never received callback from Google Voice on channel SIP/192.168.2.113:
5060-0000008f . exiting") in new stack
-- Auto fallthrough, channel 'SIP/192.168.2.113:5060-0000008d' status is
'UNKNOWN'
-- Executing [h@phone:1] GotoIf("SIP/192.168.2.113:5060-0000008d", "0?:
bridged") in new stack
-- Goto (phone,h,6)
-- Executing [h@phone:6] NoOp("SIP/192.168.2.113:5060-0000008d", "The
channel has been bridged successfully") in new stack
-- Executing [h@phone:7] Set("SIP/192.168.2.113:5060-0000008d", "GLOBAL(
gvuser)="zzzh"") in new stack
== Setting global variable 'gvuser' to '"zzz"'
-- Executing [12462562708@phone:6] NoOp("SIP/192.168.2.113:5060-0000008e
", "Never received callback from Google Voice on channel "zzzh" . exiting")
in new stack
-- Auto fallthrough, channel 'SIP/192.168.2.113:5060-0000008e' status is
'UNKNOWN'
-- Executing [h@phone:1] GotoIf("SIP/192.168.2.113:5060-0000008e", "0?:
bridged") in new stack
-- Goto (phone,h,6)
-- Executing [h@phone:6] NoOp("SIP/192.168.2.113:5060-0000008e", "The
channel has been bridged successfully") in new stack
-- Executing [h@phone:7] Set("SIP/192.168.2.113:5060-0000008e", "GLOBAL(
gvuser)="zzzh"") in new stack
== Setting global variable 'gvuser' to '"zzzh"'
[May 25 00:14:27] NOTICE[3252]: chan_sip.c:20649 handle_request_invite:
Unable to create/find SIP channel for this INVITE
-- Executing [12462550260@phone:6] NoOp("SIP/192.168.2.113:5060-0000008f
", "Never received callback from Google Voice on channel "zzzh" . exiting")
in new stack
-- Auto fallthrough, channel 'SIP/192.168.2.113:5060-0000008f' status is
'UNKNOWN'
-- Executing [h@phone:1] GotoIf("SIP/192.168.2.113:5060-0000008f", "0?:
bridged") in new stack
-- Goto (phone,h,6)
-- Executing [h@phone:6] NoOp("SIP/192.168.2.113:5060-0000008f", "The
channel has been bridged successfully") in new stack
-- Executing [h@phone:7] Set("SIP/192.168.2.113:5060-0000008f", "GLOBAL(
gvuser)="zzzh"") in new stack
== Setting global variable 'gvuser' to '"zzzh"'
debian*CLI> exit
sip.conf和extensions.conf都是那网上教程提供的，除了改了用我的GV和IPTel，
IPKall信息,改了201,202,203的密码之外没做别的改动。我的pogo linux服务器有防火
墙，
一小时之内超过6次登录，不管成功失败就屏蔽。所以服务器被侵入的可能性比较低。
谁帮忙看看怎么回事？多谢。 |
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