i**w
发帖数: 883 | 1 /etc/asterisk/scripts/callback
==============================================
#!/usr/bin/perl -w
my $option = $ARGV[0];
if ($option eq '-create') {
my $extension = $ARGV[1];
my $destination = $ARGV[2];
# write file
my $random = "$$"."$^T";
my $myfile = "/tmp/".$destination."_".$random.".call";
my $destfile = "/var/spool/asterisk/outgoing/";
open(EXTEN,">$myfile");
print EXTEN "Channel: Local/$extension-$destination\@call-back-inbound/n
\n";
print EXTEN "Contex... 阅读全帖 |
|
b*****n
发帖数: 221 | 2 用dummy教程设的RT-N16，asterisk + gv．GV可以打出,对方显示的号码也是对的.但无法打入.总是进voice mail.ipkall可以打入.
大伙帮忙看看(修正过):
gtalk.conf
[general]
context=google-a-in ; Context to dump call into
allowguest=yes
bindaddr=0.0.0.0
externip=xx.xxx.xxx.248
[guest] ; special account for options on guest account
disallow=all
allow=ulaw
context=google-a-in
[a-gtalk]
[email protected]/Talk
disallow=all
allow=ulaw
context=google-a-in
connection=a
jabber.conf
[general]
autoprune=no
autoregister=yes
[a]
t... 阅读全帖 |
|
p**i
发帖数: 688 | 3 如果你有asterisk 1.6, 建议用bridge instead of parking
关键是在outbound的context里把outbound的channel名字copy到一个global变量里,
然后在inbound的context里把inbound和outbound的channels bridge
[gv-outbound]
exten => _X.,1,Wait(1)
exten => _X.,n,Set(ACCTNAME=gvemail) ;<== Your Google Voice email
exten => _X.,n,Set(ACCTPASS=gvpassword) ;<== Your Google Voice password
exten => _X.,n,Set(RINGBACK=1NXXNXXXXXXX) ;<== Your Google Voice number with
a leading 1
exten => _X.,n,System(gvoice -e ${ACCTNAME} -p ${ACCTPASS} call ${EXTEN} ${
RI |
|
i**w
发帖数: 883 | 4 [custom-park]
exten => s,1,Wait(2)
exten => s,n,Set(GVNUM=GV_NUM)
exten => s,n,Set(CALLPARK=75)
exten => s,n,NoOp(**CALLERID: ${CALLERID(number)})
exten => s,n,GotoIf($["${CALLERID(number)}"="${GVNUM}"]?6:7)
exten => s,n,ParkedCall(${CALLPARK})
你这个里面GotoIf的index不对 |
|
i**w
发帖数: 883 | 5 extensions_dialplan.conf
=====================================
;outgoing dialplan
[outgoing]
include => parkedcalls
include => call-freephone
include => call-us-canada
include => call-china
include => call-other
include => call-internal
;dialplan for internal
[call-internal]
exten => 1000,1,Dial(SIP/1000,30)
exten => 1000,n,Congestion
exten => 1001,1,Dial(SIP/1001,30)
exten => 1001,n,Congestion
exten => 1002,1,Dial(SIP/1002,30)
exten => 1002,n,Congestion
;dialplan to call free phonenumber
[call- |
|
f*u
发帖数: 5576 | 6 Try this
[google-in]
exten => Y*****[email protected], 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?
bridged)
exten => Y*****[email protected], n, NoOp(Callerid ${CALLERID(name)})
exten => Y*****[email protected], n, Set(CALLERID(num)=${SHIFT(CALLERID(name),@)})
exten => Y*****[email protected], n, Set(CALLERID(name)=${DB(cidname/${CALLERID(
num)})})
exten => Y*****[email protected], n, Dial(SIP/101, 180, D(:1))
exten => Y*****[email protected], n(bridged),Bridge(${DB_DELETE(gv_dialout/channel
)}, p) |
|
f*u
发帖数: 5576 | 7 Try this
[google-in]
exten => Y*****[email protected], 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?
bridged)
exten => Y*****[email protected], n, NoOp(Callerid ${CALLERID(name)})
exten => Y*****[email protected], n, Set(CALLERID(num)=${SHIFT(CALLERID(name),@)})
exten => Y*****[email protected], n, Set(CALLERID(name)=${DB(cidname/${CALLERID(
num)})})
exten => Y*****[email protected], n, Dial(SIP/101, 180, D(:1))
exten => Y*****[email protected], n(bridged),Bridge(${DB_DELETE(gv_dialout/channel
)}, p) |
|
j*x
发帖数: 302 | 8 大伙看看这个.conf对吗?
[talk-numeric-outbound]
exten => _1NXXNXXXXXX,1,Dial(Gtalk/Your_GV/${EXTEN}@voice.google.com)
exten => _+1NXXNXXXXXX,1,Dial(Gtalk/Your_GV/${EXTEN}@voice.google.com)
exten => _0086X.,1,Dial(Gtalk/Your_GV/${EXTEN}@voice.google.com) |
|
k***e
发帖数: 7933 | 9 我的extension是如下， 哪里改增加delay？谢谢
exten => _[cz][email protected], 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?bridg
ed)
exten => _[a-z][email protected], n, NoOp(Callerid ${CALLERID(name)})
exten => _[a-z][email protected], n, Set(CALLERID(num)=${SHIFT(CALLERID(name),@)}
)
exten => _[a-z][email protected], n, Set(CALLERID(name)=${DB(cidname/${CALLERID(n
um)})})
exten => _[a-z][email protected], n, Dial(SIP/101, 180, aD(:1))
exten => _[a-z][email protected], n(bridged),Bridge(${DB_DELETE(gv_dialout/channe
l)}, p) |
|
d********g
发帖数: 10550 | 10 你的配置里为什么有SendDTMF？GV里没有把Call Screening关掉？
我的：
[google-in]
exten => , 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?bridged)
exten => , n, NoOp(Callerid ${CALLERID(name)})
exten => , n, Set(CALLERID(num)=${SHIFT(CALLERID(name),@)})
exten => , n, Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten => , n, Dial(SIP/, 180, D(:1))
exten => , n(bridged),Bridge(${DB_DELETE(gv_dialout/channel)}, p)
我觉得问题应该就在这个DTMF。发... 阅读全帖 |
|
g**d
发帖数: 723 | 11 哈哈, 把kanke的楼歪掉了, 重开一个问一下.用了如下的conf, GV_NUMBER也在nonoh里
verify了. 但是打在我手机上仍然显示"private".怎么解决?
asterisk1.6
extension.conf
[call-with-nonoh]
exten => _X.,1,ExecIf($[ "${CALLERID(name)}" = "" ] ,Set(CALLERID(name)=1${
GV_NUMBER}))
exten => _X.,n,Set(CALLERID(number)=1${GV_NUMBER})
exten => _X.,n,Set(CALLERPRES()=allowed_not_screened)
exten => _X.,n,Dial(SIP/${EXTEN}@nonoh,90,r)
sip.conf
[nonoh]
type=peer
host=sip.nonoh.net
username=NONOH_ID
secret=NONOH_PASS
nat=yes
fromdomain=sip.nonoh.net
fromuser=1 |
|
C*******1
发帖数: 422 | 12 In debug messages, I can see DTMF(1) got sent out. But I guess it is sent
out immediately after 101 was picked up - even before GV said "press 1
...".
The problem has been solved by replacing
exten => k*******[email protected], n, Dial(SIP/101,180,D(:1))
with
exten => m************[email protected], n, Answer
exten => m************[email protected], n, Wait(1)
exten => m************[email protected], n, SendDTMF(1)
exten => m************[email protected], n, Dial(SIP/101,180,) |
|
s*******d
发帖数: 4135 | 13 他说搞定了：
The problem has been solved by replacing
exten => k*******[email protected], n, Dial(SIP/101,180,D(:1))
with
exten => m************[email protected], n, Answer
exten => m************[email protected], n, Wait(1)
exten => m************[email protected], n, SendDTMF(1)
exten => m************[email protected], n, Dial(SIP/101,180,) |
|
C*******1
发帖数: 422 | 14 In debug messages, I can see DTMF(1) got sent out. But I guess it is sent
out immediately after 101 was picked up - even before GV said "press 1
...".
The problem has been solved by replacing
exten => k*******[email protected], n, Dial(SIP/101,180,D(:1))
with
exten => m************[email protected], n, Answer
exten => m************[email protected], n, Wait(1)
exten => m************[email protected], n, SendDTMF(1)
exten => m************[email protected], n, Dial(SIP/101,180,) |
|
s*******d
发帖数: 4135 | 15 他说搞定了：
The problem has been solved by replacing
exten => k*******[email protected], n, Dial(SIP/101,180,D(:1))
with
exten => m************[email protected], n, Answer
exten => m************[email protected], n, Wait(1)
exten => m************[email protected], n, SendDTMF(1)
exten => m************[email protected], n, Dial(SIP/101,180,) |
|
h*******a
发帖数: 3279 | 16 谢谢a大 我的是spa1001 版上买的 简单的填了ip地址和用户名密码 现在可以打出了
但是怎么样都接不了电话
已经参照这一段改了extensions.conf
The problem has been solved by replacing
exten => k*******[email protected], n, Dial(SIP/101,180,D(:1))
with
exten => m************[email protected], n, Answer
exten => m************[email protected], n, Wait(1)
exten => m************[email protected], n, SendDTMF(1)
exten => m************[email protected], n, Dial(SIP/101,180,)
不过有电话打进来的时候还是啥反应没有恩……
改这段之前在电脑的softphone里倒是能接的 改了之后哪里都接不到了……
不知道还有哪里需要改的吗？谢谢～～ |
|
F******k
发帖数: 7375 | 17 Do the following 3 steps:
1. Open extensions.conf, replace
exten => m************[email protected], n, Dial(SIP/101,180,D(:1))
with
exten => m************[email protected], n, Answer
exten => m************[email protected], n, Wait(1)
exten => m************[email protected], n, SendDTMF(1)
exten => m************[email protected], n, Dial(SIP/101,180,)
NOTE: use your own gmail account instead of m************[email protected]!!!!!!!!
!!!!!!!!!!!!!!!!!!!!!!
2. Open modules.conf, add the following:
load => app_senddtmf.so ;Send ... 阅读全帖 |
|
a9
发帖数: 21638 | 18 exten => 101, 1, Dial(SIP/101, 10)
exten => 102, 1, Dial(SIP/102, 10)
exten => 103, 1, Dial(SIP/103, 10)
exten => 104, 1, Dial(SIP/104, 10)
exten => 105, 1, Dial(SIP/105, 10)
是不是这个10改成30？ |
|
s**n
发帖数: 449 | 19 i use:
exten => g*[email protected], n, Dial(SIP/101, 180, D(:1))
instead of your:
exten => g*[email protected], n, Answer
exten => g*[email protected], n, Wait(1)
exten => g*[email protected], n, SendDTMF(1)
exten => g*[email protected], n, Dial(SIP/101,180,)
will try your solution tonight.
thanks.
})
, |
|
l********l
发帖数: 9452 | 20
replace
exten => m************[email protected], n, Dial(SIP/101,180,D(:1))
with
exten => m************[email protected], n, Answer
exten => m************[email protected], n, Wait(1)
exten => m************[email protected], n, SendDTMF(1)
exten => m************[email protected], n, Dial(SIP/101,180,) |
|
s*******d
发帖数: 4135 | 21 我的电话这2天也出现这个问题了，考古了一下。按下面的方法就可以了。
发信人: liulinglll (liuliu), 信区: voip
标 题: Re: asterisk gtalk接不起来的，你们有没有试过
发信站: BBS 未名空间站 (Fri Dec 9 00:22:39 2011, 美东)
replace
exten => m************[email protected], n, Dial(SIP/101,180,D(:1))
with
exten => m************[email protected], n, Answer
exten => m************[email protected], n, Wait(1)
exten => m************[email protected], n, SendDTMF(1)
exten => m************[email protected], n, Dial(SIP/101,180,) |
|
s*******d
发帖数: 4135 | 22 我的电话这2天也出现这个问题了，考古了一下。按下面的方法就可以了。
发信人: liulinglll (liuliu), 信区: voip
标 题: Re: asterisk gtalk接不起来的，你们有没有试过
发信站: BBS 未名空间站 (Fri Dec 9 00:22:39 2011, 美东)
replace
exten => m************[email protected], n, Dial(SIP/101,180,D(:1))
with
exten => m************[email protected], n, Answer
exten => m************[email protected], n, Wait(2)
exten => m************[email protected], n, SendDTMF(1)
exten => m************[email protected], n, Dial(SIP/101,180,)
exactly,
( |
|
m*d
发帖数: 7658 | 23 incoming call铃声能响，但是接不起来
前两天还能接起来
于是把D(:1)换成了SendDTMF
replace
exten => m************[email protected], n, Dial(SIP/101,180,D(:1))
with
exten => m************[email protected], n, Answer
exten => m************[email protected], n, Wait(1)
exten => m************[email protected], n, SendDTMF(1)
exten => m************[email protected], n, Dial(SIP/101,180,)
但是遇到错误
[Mar 31 13:07:37] ERROR[4827]: pbx.c:3652 ast_func_read: Function DB_EXISTS
not registered
[Mar 31 13:07:37] ERROR[4827]: pbx.c:3652 ast_func_read: function DB not
re... 阅读全帖 |
|
m*d
发帖数: 7658 | 24 比如配置101，102，103三个extension，每个extension用自己单独的gv打出接入
101，102，103之间用分机号可以互通，但是用gv的时候互不影响
分机互通的是这样
[internal-office]
exten => _1XX,1,answer()
exten => _1XX,n,dial(sip/${EXTEN})
exten => _1XX,n,Wait(15)
exten => _1XX,n,hangup()
各个extension应该怎么配置？
这样的extensions.conf应该怎么写？ |
|
c**v
发帖数: 103 | 25 装的最简单的asterisk1.4,没有装pbx-in-a-flash;
现在打进（sipgate->pbx)，打出(pbx->sipgate->destination)都设好了;
想把google voice callback 加上遇到了问题，
不知道 [from-trunk]下是怎么定义的？
I installed pygooglevoice following instruction in doc here:
http://code.google.com/p/pygooglevoice/,
It generated "extensions_custom.conf" (I only used variable to replace
the sensitive information):
[custom-gv]
exten => _X.,1,Wait(1)
exten => _X.,n,Set(ACCTNAME=${GACCNAME})
exten => _X.,n,Set(ACCTPASS=${GACCSECRET})
exten => _X.,n,Set(RINGBACK=${S |
|
k****t
发帖数: 2288 | 26 520gu是没有办法装gui的，跑不起来吧！
1）我是这么解决的，在spa3000的dial plan中将第一个1屏蔽掉，给asterisk的还是只
有10位。
2）在/opt/etc/asterisk/下修改两个文件extensions.conf和sip.conf。我是用的voip
discount做例子。
1，extensions.conf， _011X那行是新加的。
[from-internal]
include => parkedcalls
exten => _011X.,1,Dial(SIP/${EXTEN}@voipdiscount)
exten => _NXXNXXXXXX,1,NoOp()
exten => _NXXNXXXXXX,n,Wait(1)
2，sip.conf，增加下面的。【voipdiscount】这部分是copy 上面sipgate的，我只是修
改了一下account name和password以及proxy。
[voipdiscount]
disallow=all
username=voipdiscount account name
type= |
|
a9
发帖数: 21638 | 27 首先，注意两个
1是context=
2是[email protected]/from-Gizmo5里这个from-Gizmo5
在extensions.conf里，电话来的时候，先找[context]这段，注意context要替换成1里
的context=后面的内容
然后就开始找exten=from-Gizmo5这样的段。
这里又有几点
(1)这里一般用数字，比较好匹配。比如用你gizmo5的号码
(2)需要前面加一个下划线。
（3）找到后从序号1开始。
举个例子。你的context是fromgizmo，register斜线后面是17471234567，就应该这么写
[fromgizmo]
exten => _17471234567,1,NoOp(incoming call from ${CALLERID(num)})
exten => _17471234567,n,Dial(SIP/102)
这样，来电话后，第一步，是显示一个信息，第二步，就拨通102分机。
这就是最简单的一个exten，asterisk有很多功能，你可以加进去。
Call from ' |
|
a9
发帖数: 21638 | 28 exten => _9001,1,NoOp(call in)
exten => _9001,n,Answer()
exten => _9001,n,Read(inputed,,,,,5)
exten => _9001,n,Dial(SIP/${inputed}@onesuite,30,tT) |
|
g**d
发帖数: 723 | 29 ianw 你这两个pattern有重复的， 我打01186xxx, call-china pick it up and dail
00861186xxx@nonoh. 大概取决于asterisk存储的顺序
[call-other]
exten => _011ZX.,1,Goto(call-with-nonoh,00${EXTEN:3},1)
;[call-china]
;exten => _0ZX.,1,Goto(call-with-nonoh,0086${EXTEN:1},1) |
|
k******t
发帖数: 163 | 30 group效果很好。正常通话可以工作，我方没有拨通也工作。
exten => _00XX.,1,Set(GROUP(E001)=OUTBOUND_GROUP_ONCE)
exten => _00XX.,n,GotoIf($[${GROUP_COUNT(OUTBOUND_GROUP_ONCE@${E001})} > 1]?
BLOCK)
exten => _00XX.,n,Dial(SIP/${EXTEN}@localphone) |
|
k******t
发帖数: 163 | 31 allow=ulaw
g729无法工作。（CPU太差）
[myDISA]
exten => s,1,Answer()
exten => s,n,Read(OUTNUM)
exten => s,n,Goto(dial-800,${OUTNUM},1)
exten => s,n(end),Hangup
如果由外线拨入GIZMO工作正常：
-- Executing [s@myDISA:1] Answer("SIP/Gizmo1-00000003", "") in new stack
-- Executing [s@myDISA:2] Read("SIP/Gizmo1-00000003", "OUTNUM") in new
stack
-- User entered '005388888'
-- Executing [s@myDISA:3] Goto("SIP/Gizmo1-00000003", "dial-800,
005388888,1") in new stack
-- Goto (dial-800,005388888,1)
-- Exec |
|
r****t
发帖数: 10904 | 32 这个问题(1.6 bridge 设置)是 piii 帮我找出来解决了的, 可以翻前面我的帖子。
我的从 ata 打出的 context:
[from-ata]
exten => _NXXNXXXXXX,1,Dial(Local/1${EXTEN}@outbound-gv-sipgate/n)
exten => _1NXXNXXXXXX,s,Dial(Local/${EXTEN}@outbound-gv-sipgate/n)
我的 [outbound-gv-sipgate] 类似你的 [gv-outbound],
我的 [from-sipgate] 类似你的 [gv-inbound]
这里是 log, 注意 bridge 是和 Local channel bridge
-- Executing [1xxxxxxxxxx@from-sipgate:8] Bridge("SIP/sipgate-
00000214", "Local/1123xxxxxxx@outbound-gv-sipgate-c8a3;2") in new stack
-- Local/1123xxxxxx... 阅读全帖 |
|
r****t
发帖数: 10904 | 33 不用动 feature。
你的 log 和我基本完全一样，看不出什么问题 :(
-- Executing [408xxxxxxx@gv-outbound:3] Set("Local/408xxxxxxx@gv-
outbound-02a7;2", "DB(gv_dialout/channel)=Local/408xxxxxxx@gv-outbound-
02a7;2 ") in new stack
-- Executing [6504xxxxxxx@ext-did:3] Bridge("SIP/SipGate-0000000d",
"Local/408xxxxxxx@gv-outbound-02a7;2 ") in new stack
这个 warning 应该是 cannot get its lock 了，要不重启 asterisk 试试, 不应该啊。
[2010-10-26 15:18:52] WARNING[4779]: features.c:4649 bridge_exec: Bridge
failed because channel Local/408xxxxx... 阅读全帖 |
|
a9
发帖数: 21638 | 34 首先asterisk 1.8在dockstar上运行不正常
gtalk+asterisk 1.6在dockstar上也跑的不太正常。
安装asterisk.首先安装
iksemel
wget http://iksemel.googlecode.com/files/iksemel-1.4.tar.gz
解压 tar -zxvf iksemel-1.4.tar.gz并进入这个目录
配置 ./configure --prefix=/usr (不装在usr下好像会有问题)
编译安装 make & make install
下载编译asterisk 1.8
./configure --use-ssl=/usr/lib 注意看usr/lib下有没有libssl*.so
make menuselect 选上channels->gtalk,resources->res_jabber
编译安装 make & make install
配置
/etc/asterisk/jabber.conf
[general]
debug=no
autoprune=no
autoregister=... 阅读全帖 |
|
|
|
y******g
发帖数: 120 | 37 记得 nonoh 用 00 而不是 011 做国际区号开头。我只找到楼主在以下设置了 01186
用 nonoh 打出。楼主在什么地方替换011为00呢？ 还是不需要？我的电话簿都用的
0086，不想改到01186 了。试了下，照以下设置，01186可以打，0086打不通。
[custom-international]
exten => _01186X.,1,Set(CALLERID(num)=${1XXXX#})
exten => _01186X.,n,Dial(SIP/${EXTEN}@nonoh,50,trg)
exten => _01186X.,n,Hangup
some
directly, |
|
y******g
发帖数: 120 | 38 记得 nonoh 用 00 而不是 011 做国际区号开头。我只找到楼主在以下设置了 01186
用 nonoh 打出。楼主在什么地方替换011为00呢？ 还是不需要？我的电话簿都用的
0086，不想改到01186 了。试了下，照以下设置，01186可以打，0086打不通。
[custom-international]
exten => _01186X.,1,Set(CALLERID(num)=${1XXXX#})
exten => _01186X.,n,Dial(SIP/${EXTEN}@nonoh,50,trg)
exten => _01186X.,n,Hangup
some
directly, |
|
i**w
发帖数: 883 | 39 原因就在下面的第一行
========================================================
exten => m************[email protected], n, Answer
exten => m************[email protected], n, Wait(1)
exten => m************[email protected], n, SendDTMF(1)
exten => m************[email protected], n, Dial(SIP/101,180,)
======================================================== |
|
x*h
发帖数: 757 | 40 我猜应该是的，请问可以给个sample的设置吗？
我现在的设置应该是，
[talk-numeric-outbound]
exten => _1NXXNXXXXXX,1,Dial(Gtalk／asterisk/${EXTEN}@voice.google.com)
exten => _+1NXXNXXXXXX,1,Dial(Gtalk/asterisk/${EXTEN}@voice.google.com)
这里ms也没有给出international call的方案，谢谢了！
https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
|
m**v
发帖数: 373 | 41 有个问题，别人打我的GV，我的phonerlite（接在asteriskshang)能够接电话，如果对
方先挂电话，asterisk就会有个warning：
WARNING[1094]: features.c:5845 bridge_exec: Bridge failed because channel d
oes not exists or we cannot get its lock
是不是跟下面的extensions.conf里面的两行有关？ 这两行什么意思呢？
exten => d***[email protected], 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?bridged
)
exten => _[a-z][email protected], n(bridged),Bridge(${DB_DELETE(gv_dialout/channe
l)}, p)
有的时候挂电话asterisk干脆就死了，有个问题，别人打我的GV，我的phonerlite（接
在asteriskshang)能够接电话，如果对
方先挂电话，asterisk就会有个warnin... 阅读全帖 |
|
e******o
发帖数: 1160 | 42 呵呵，提醒我了，确实是这个地方出错，但是错误更正如下：
exten => _01186X.,1,Set(CALLERID(num)=xxxxxx)
exten => _01186X.,n,Dial(SIP/${EXTEN}@nonoh,50,trg)
exten => _01186X.,n,Hangup |
|
l*******e
发帖数: 3584 | 43 接着求助，我在CONF里加了这么一句
[google-international]
exten => _01186XXXXXXXXXXX,1,Dial(Gtalk/#####/${EXTEN}@voice.google.com)
exten => _01186XXXXXXXXXX,1,Dial(Gtalk/#####/${EXTEN}@voice.google.com)
觉得应该可以了，可是还是不行，求助 |
|
h**c
发帖数: 118 | 44 大家好，
我现在是在opt工作，正式有paid。 现在问题就是我opt是今年3月底过期。1月回国了
一个月耽误了很多opt 延期的申请，现在学校刚吧i20弄出来，还没寄到我这里，我即
使开始opt extens申请也的下个星期了。这样也就还有6周时间，我觉得根据以往经验
，这个时间肯定不够用（悲观点来说）。但我如果申请今年h1b，假设我人品大爆发抽
中了，由于我是在opt过期之后才申请的h1b，如果抽中，我从出结果的4月份到10月份
h1b生效之前不能工作（correct me if I was wrong）.那问题来了，我现在有两个方案
1。 今年不抽h1b了，等先拿到opt extens再申请，缺点就是只有一次抽签机会了
2. 今年也申请h1b， 同时也申请opt extens, 缺点是如果抽中如上文所说，我4-10月
不能工作。
跪求各位大神仙。能否给小弟指一条明路。
多谢了！ |
|
c******y
发帖数: 97 | 45 我自己觉得自己的CASE不是很强，以前列出过：没奖没报道，15 papers, 186
citations, 7 reviews, 1 international science fair judge。
当初也有人和说我一切都会顺利，我说他们不懂我的事。但今天走到这一步，我也想鼓
励一些还在奋斗的人们，一切都会好的，坚持。
折腾这几年的所有I-140和H1B。
WAC08 6/25/200 I129
WAC08 9/25/200 I129
WAC12 6/11/201 I129
WAC10 7/26/201 I129
EAC06 1/6/200 I129
WAC11 5/11/201 I129
WAC08 4/24/200 I129
LIN12 3/27/201 I140
SRC12 6/11/201 I140
LIN10 7/18/201 I140
再看看这些年折腾的MILESTONE，有歇着的时候吗？
Date Mat... 阅读全帖 |
|
k******t
发帖数: 163 | 46 repast兄：
我想学习你的方法，
[dial-out-context]
exten => _NXXNXXXXXX,1,Dial(local/${EXTEN}@gv-outbound/n)
[gv-outbound]
exten => _NXXNXXXXXX,1,NoOp
。。。。。。。。。。。。。。
但是：
-- Executing [8885285258@dial-out-context:1] Dial("SIP/1000-00000006", "
local/8885285258@gv-outbound/n") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/1000-00000006' status is 'CHANUNAVAIL'
请教如何是好？ |
|
p**i
发帖数: 688 | 47 你试试下面这个
[from-internal]
include => parkedcalls
exten => _011X.,1,Set(CALLERID(num)=1NXXNXXXXXX) ; the number I set as my
caller id in softphone (my gv number)
exten => _011X.,n,Dial(SIP/${EXTEN}@nonoh) |
|
g**d
发帖数: 723 | 48 我发现Bridge很不好用啊, 在gv_dialout后要wait(),这个时间晚了太慢, 早了线断了.
exten => _X.,n,Set(DB(gv_dialout/channel)=${GV_CALLCHAN})
exten => _X.,n,Wait(3)
exten => _X.,n,Hangup
反正我搞了半天发现没有park()用在这里合适. 是我用错了么? |
|
k******t
发帖数: 163 | 49 exten => _00XX.,n,Dial(SIP/${EXTEN:2}@localphone,30)
成功，可以打电话。
可是Hangup后
exten => _00XX.,n,Set(GLOBAL(ZEROONEONE)="NotInUse")
不执行。 |
|
