m*d
发帖数: 7658 | 1 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 |
|
a9
发帖数: 21638 | 2 extensions.conf
[from-sip-internal]
exten => _XXXXXXXXXX,1,Dial(gtalk/g1/+1${EXTEN}@voice.google.com)
exten => _1XXXXXXXXXX,1,Dial(gtalk/g1/+${EXTEN}@voice.google.com)
[gtalkin]
exten => h,1,NoOp(CALL TO ${EXTEN} hangup)
exten => h,n,Hangup()
exten => [email protected],1,NoOp(CALL FROM ${CALLERID(num) (${CALLERID(na
me)})} THROUGH GTALK LINE)
exten => [email protected],n,Answer()
exten => [email protected],n,Wait(3)
exten => [email protected],n,SendDTMF(1)
exten => [email protected],n,Dial(SIP/8001,30,tT)
exten => [email protected]... 阅读全帖 |
|
a9
发帖数: 21638 | 3 extensions.conf
[from-sip-internal]
exten => _XXXXXXXXXX,1,Dial(gtalk/g1/+1${EXTEN}@voice.google.com)
exten => _1XXXXXXXXXX,1,Dial(gtalk/g1/+${EXTEN}@voice.google.com)
[gtalkin]
exten => h,1,NoOp(CALL TO ${EXTEN} hangup)
exten => h,n,Hangup()
exten => [email protected],1,NoOp(CALL FROM ${CALLERID(num) (${CALLERID(na
me)})} THROUGH GTALK LINE)
exten => [email protected],n,Answer()
exten => [email protected],n,Wait(3)
exten => [email protected],n,SendDTMF(1)
exten => [email protected],n,Dial(SIP/8001,30,tT)
exten => [email protected]... 阅读全帖 |
|
a9
发帖数: 21638 | 4 这个bug我觉得可能是这样：
用xmpp连上去的时候，你一接电话，他会说按1接听，按2转留言什么的。
这时候要按1就听到了。
然而好像这个模块的发送dtmf有问题，你用电话机按的1传不过去。
就只好在asterisk里面senddtmf(1)，这样如果senddtmf太早了，那边就收不到了。
不过这好像是很早以前看到的一片文章，现在不知道还是不是这个情况。
也接 |
|
F******k
发帖数: 7375 | 5 到底应该是D(:1)还是SendDTMF(1)？刚刚看了俺的，是D(:1),但有时候会接不起来。要
是改成wait and SendDTMF(1),具体怎么改？谢谢 |
|
r****t
发帖数: 10904 | 6 觉得 a9 说的是 SendDTMF(1) 不一定能行，需要不断 SendDTMF(1) 才行。
没试过，我还在 1.6+pygooglevoice |
|
|
C*******1
发帖数: 422 | 8 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 | 9 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 | 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,) |
|
|
i**w
发帖数: 883 | 15 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,)
======================================================== |
|
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～～ |
|
p**i
发帖数: 688 | 17 这个forward到google chat的选项是不是2011年才出现的?以前从来没注意过
It is the best choice so far (sipgate或ipkall可以暂时休息了)!
我的context for in-bound calls (change EXT1-3 to the desired extensions to
ring)
[gtalk-in]
exten = s,1,NoOp(${CHANNEL})
exten = s,n,Set(CALLERID(name)=${CUT(CHANNEL,/,1)})
exten = s,n,Set(MYCID=${CUT(CHANNEL,+,2)})
exten = s,n,Set(CALLERID(num)=${CUT(MYCID,-,1)})
exten = s,n,Wait(1)
exten = s,n,SendDTMF(1)
exten = s,n,Dial(SIP/EXT1&SIP/EXT2&SIP/EXT3)
exten = s,n,Hangup()
那个 |
|
p**i
发帖数: 688 | 18 我这个google帐号是专门留给google voice/chat用的, 应该不会有其他login, 除非被
hack了
为了适应GV回拨, context修改如下
[gtalk-in]
exten => s,1,NoOp(${CHANNEL})
exten =>s,n,Set(CALLERID(name)=${CUT(CHANNEL,/,1)})
exten =>s,n,Set(MYCID=${CUT(CHANNEL,+,2)})
exten =>s,n,Set(CALLERID(num)=${CUT(MYCID,-,1)})
exten =>s,n,Set(DID_CID=${CALLERID(num)})
exten => s,n,Wait(1)
exten => s,n,SendDTMF(1)
exten => s,n,GotoIf($[${DID_CID:1} = XXXXXXXXXX]?:normalcall)
;<==10-digit GoogleVoice number
exten => s,n,NoCDR()
exten =>s,n,Bridge(${DB_DELET... 阅读全帖 |
|
t***n
发帖数: 546 | 19 鉴于有人要求详细配置，现在把几个配置文件贴出来，欢迎大牛们指正
gtalk.conf***********************
[general]
context=google-in ; Context to dump call into
allowguest=yes
[guest] ; special account for options on guest account
disallow=all
allow=ulaw
context=google-in
[dummy-gtalk]
username=d***[email protected]
disallow=all
allow=ulaw
context=google-in
connection=dummy
[user3-gtalk]
username=u***[email protected]
disallow=all
allow=ulaw
context=google-in
connection=user3
[user1-gtalk]
username=u***[email protected]
disallow=all
... 阅读全帖 |
|
F******k
发帖数: 7375 | 20 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 |
|
z****n
发帖数: 26 | 21 多谢a9大大指点,我的extensions.conf见下,应该改哪里,麻烦大师指点一下,谢谢!
[general]
static=yes
writeprotect=no
clearglobalvars=no
allowguest=no
alwaysauthreject=yes
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/G2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
[default]
exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten => s,n,Dial(SIP/101,10)
ext... 阅读全帖 |
|
|
j**********n
发帖数: 527 | 23 对，我也遇到这个问题。load了senddtmf的module就好了。 |
|
s*****e
发帖数: 404 | 24 怎么load senddtmf的module呀？要加到module.conf里吗？在moduls.conf里，autoload=yes。谢谢。 |
|
l****n
发帖数: 3081 | 25 请教具体怎么加?加了还用不用load那个senddtmf了?
我倒是能打能接,就是接的时候要等电话响三声再接,不然就是听不到对方,对方还是不
停的拨号音. |
|
k****t
发帖数: 2288 | 26 装了1.8,也用了google talk，但是也用了sipgate来接sipgate的电话或者是用来打800
电话。但是现在我从sipgate的另外一个号码call这个650XXXXX2的电话得到如下的log
，给人的感觉就是进了outbound的了，好奇怪；看了一下externsion。conf，应该是
come in phone call会自动dial default的呀？
其他的都很正常：从1001往外打电话，接gtalk来的电话，但是没有办法接从sipgate过
来的电话。
== Using SIP RTP CoS mark 5
-- Executing [650XXXXXX2@outbound:1] Set("SIP/SipGate-0000001d", "
CALLERID(dnid)=1650XXXXXX2") in new stack
-- Executing [650XXXXXX2@outbound:2] Goto("SIP/SipGate-0000001d", "
1650XXXXXX2,1") in new stack
-- Goto (o... 阅读全帖 |
|
r*****8
发帖数: 2697 | 27 GV中call screening设置off以后, 我的obi和asterisk都没有你说的这个问题, 至少到
现在为止还没有碰到过, 建议你申请一个新的gmail和GV试试
另外, 请参考我的extensions.conf中的
[google-in]
exten => g*[email protected], 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?bridged)
exten => g*[email protected], n, NoOp(Callerid ${CALLERID(name)})
exten => g*[email protected], n, Set(CALLERID(num)=${SHIFT(CALLERID(name),@)})
exten => g*[email protected], n, Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})
})
exten => g*[email protected], n, Answer
exten => g*[email protected], n, Wait(1)
exten => g*2@g... 阅读全帖 |
|
s**n
发帖数: 449 | 28 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.
})
, |
|
a9
发帖数: 21638 | 29 你们这些怎么还都有gotoif检查数据库呢？这是用ipkall转接的时候才用的吧？
你这个要加，可以在NoOp前面加上， Answer........ delay ...... SendDTMF(1).....
bridg
)}
(n
channe |
|
b*****n
发帖数: 221 | 30 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 |
|
s**n
发帖数: 449 | 31 nod, i changed from D(:1) to wait and SendDTMF(1)
1) |
|
l********l
发帖数: 9452 | 32
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**n
发帖数: 449 | 33 原来用的是 asterisk1.8 + gv + pap2，基本work ok, 有点小问题。
1. 如果计算机上login了gmail, pap2上的电话就不响了。
2. 接听时，有时候需要按1, 即使有senddtmf(1)
用了obihai100, 完全没有以上问题。 |
|
e******o
发帖数: 1160 | 34 用的dockstar，装的archlinux，安装的asterisk 1.8，很是奇怪，打中国电话老是出错
。。。。。
dockstar*CLI> sip show peers
Name/username Host Dyn Force
rport ACL Port Status
1001/1001 192.168.1.138 D N
A 5060 OK (215 ms)
1002 (Unspecified) D N
0 Unmonitored
SipGate/xxx 204.155.28.10 N
5060 OK (24 ms)
nonoh/bbb... 阅读全帖 |
|
a9
发帖数: 21638 | 35 需要按1吗？
我这儿一直不行。不管用Answer+senddtmf还是Dial(D(1)) |
|
c**s
发帖数: 771 | 36 en, 能响.
But 3 or 4 times out of 10, there is dead air when I pick up the phone on
incoming calls. Yes, it used to work fine, but started to have this issue
some time last year. I thought it was only me. Now from the link you
provided, I know it is a problem to many people.
Somebody in that link said they had a plausible solution by adding a short
sound before sending DTMF
Answer(1)
Playback(hello-world)
SendDTMF(1)
I am testing this out. |
|
a9
发帖数: 21638 | 37 你*上设置的都是你ata的格式。如果是gtalk模块，他往gooogle voice用senddtmf和
dial()命令发的时候，总是以rfc2833发的。但是，他这个gtalk模块有bug，我没仔细
研究出来是哪儿。
可能是在发送dtmf的时候，sip端还没有数据，因此，rtp头里要求的seqno和timespan
都是不正确的，发给google voice就不对了。
我建议你试试我写的程序。我会一直不停的发dtmf，一直到google voice能检测到。 |
|
m*d
发帖数: 7658 | 38 我的extensions.conf里面设置如下，打死接电话都不响，谁给看看
另外如何debug asterisk，看到执行到那一行了呢？
我试了asterisk -r 20多个v -d，也没有看到相关的显示，如果知道asterisk收到了
Google voice的request了呢？
[google-in]
exten => , 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?bri
dged)
exten => , n, NoOp(Callerid ${CALLERID(name)})
exten => , n, Set(CALLERID(num)=${SHIFT(CALLERID(name),@)
})
exten => , n, Set(CALLERID(name)=${DB(cidname/${CALLERID(
num)})})
exten => ... 阅读全帖 |
|
d********g
发帖数: 10550 | 39 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(${DB_DELETE(gv_dialout/channel)}, p)
我觉得问题应该就在这个DTMF。发... 阅读全帖 |
|
s*******d
发帖数: 4135 | 40 我的电话这2天也出现这个问题了，考古了一下。按下面的方法就可以了。
发信人: liulinglll (liuliu), 信区: voip
标 题: Re: asterisk gtalk接不起来的，你们有没有试过
发信站: BBS 未名空间站 (Fri Dec 9 00:22:39 2011, 美东)
replace
exten => m************[email protected], n, Dial(SIP/101,180,D(:1))
with
exten => m************[email protected], n, Answer
exten => m************[email protected], n, Wait(1)
exten => m************[email protected], n, SendDTMF(1)
exten => m************[email protected], n, Dial(SIP/101,180,) |
|
s*******d
发帖数: 4135 | 41 我的电话这2天也出现这个问题了，考古了一下。按下面的方法就可以了。
发信人: liulinglll (liuliu), 信区: voip
标 题: Re: asterisk gtalk接不起来的，你们有没有试过
发信站: BBS 未名空间站 (Fri Dec 9 00:22:39 2011, 美东)
replace
exten => m************[email protected], n, Dial(SIP/101,180,D(:1))
with
exten => m************[email protected], n, Answer
exten => m************[email protected], n, Wait(2)
exten => m************[email protected], n, SendDTMF(1)
exten => m************[email protected], n, Dial(SIP/101,180,)
exactly,
( |
|
k****t
发帖数: 2288 | 42 我的是1.8用的是gtalk。一直很好~~~
extern => x*[email protected] ,n, Answer
extern => x*[email protected] ,n, Wait(2)
extern => x*[email protected] ,n, SendDTMF(1)
extern => x*[email protected] ,n, Dial(SIP/1000,180, ) |
|
m******t
发帖数: 4077 | 43 extensions.conf
----------------------
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/G2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
[default]
exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten => s,n,Dial(SIP/101,10)
exten => s,n, Hangup
exten => 101, 1, Dial(SIP/... 阅读全帖 |
|
|
a*****s
发帖数: 8146 | 45 贴上我的google in看看和你的一不一样.
[google-in]
exten => [email protected], 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?bridged)
exten => [email protected], n, NoOp(Callerid ${CALLERID(name)})
exten => [email protected], n, Set(CALLERID(num)=${SHIFT(CALLERID(name),@)})
exten => [email protected], n, Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten => [email protected], n, Answer
exten => [email protected], n, Wait(1)
exten => [email protected], n, SendDTMF(1)
exten => [email protected], n, Dial(SIP/101,180,)
exten => [email protected], n(brid... 阅读全帖 |
|
a9
发帖数: 21638 | 46 你用的obi?
obi是先接起来按1，再响你的座机。这样gv就认为你已经接起电话了。
其实asterisk也一样。你那样Answer,SendDTMF(1)，就已经接起电话了。 |
|
m*d
发帖数: 7658 | 47 不过以前asterisk work的时候也senddtmf了，手机一样也能响啊 |
|
a9
发帖数: 21638 | 48 不太可能。只要成功接起来了(senddtmf成功），另外的转移就不响了呀。 |
|
w*m
发帖数: 1806 | 49 dialplan,
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
IAXINFO=guest ; IAXtel username/
password
TRUNK=Zap/G2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (
usually 1 or 0)
[default]
exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten => s,n,Dial(SIP/103,10... 阅读全帖 |
|
