t***n 发帖数: 546 | 1 最近折腾神由上的astersik不少时间,总算让多google voice帐号和SipGate能正常运
行。分享一下自己
的心得,欢迎大牛们拍砖。
多google voice主要用于打出,打入的话就都forward到一个SipGate号码就行了。好处:
1. 多个人用同一个电话时,可以在电话号码上加前缀决定从谁的google voice走。也
决定了对方的来电
显
示显示的是谁的电话。
2. 当某人有自己的专属电话时,可以默认所有电话都从他自己的google voice走。
假定有一下3个GV 帐号:
d***[email protected] 默认打出打入号码,相当于一个landline。 只forward到google
chat
u***[email protected], u***[email protected] 两个人的常用google voice号码
1. 先配置gtalk.conf和jabber.conf,把3个帐号都连上。参考
http://www.mitbbs.com/clubarticle_t1/voip/31214573_0_1.html
2. 配置sip.conf, 注意公共电话的context=outbound_Common, 各自的私人电话则设为
context=outbound_user1 或者 context=outbound_user2.
3. 配置extensions.conf
现在global 里面定义一个变量 GoogleOut=dummy, 默认用dummy帐号打出
然后加入context outbound_user1
[outbound_User1] ; User1's phone, using User1's google voice out
exten => _!,1,Set(GoogleOut=User1)
exten => _!,n,Goto(outbound,${EXTEN},1)
把默认的GoogleOut变量赋值为user1的名字。别忘了把outbound里面所有用到google
voice的地方的
google 用户名都用变量GoogleOut替换。 这样user1就设好了。从user1的私人电话打
出来的都会自动
把google的用户名转为user1,然后走outbound正常打出。user2的设置类似
接下来就是设置[outbound_Common] 了,在共用的电话上通过加前缀来区分从哪个
google voice帐号
走。
[outbound_Common] ;Public phones, need to distinguish ID from the dialing
pattern
exten => _001.,1,Goto(outbound_user1,${EXTEN:3},1)
exten => _002.,1,Goto(outbound_user2,${EXTEN:3},1)
; Begin normal calling using default google voice out, if no ID prefix is
added
exten => _X!,1,Goto(outbound,${EXTEN},1)
exten => _[a-z]!,1,Goto(outbound,${EXTEN},1)
exten => _[A-Z]!,1,Goto(outbound,${EXTEN},1)
如果前缀是001,就走outbound_user1,并且把号码的前3位(001)去掉。
如果前缀是002,就走outbound_user2,并且把号码的前3位(002)去掉。
如果前缀不match,就直接走outbound普通路线,从dummy的google voice打出。 | D*******l 发帖数: 5462 | | g****8 发帖数: 2828 | | k***e 发帖数: 7933 | 4 赞一下!
你是用lenny还是squeeze?asterisk 1.8吗? 好像多个GV不能forward到同一个
sipgate吧?
处:
【在 t***n 的大作中提到】![](/moin_static193/solenoid/img/up.png) : 最近折腾神由上的astersik不少时间,总算让多google voice帐号和SipGate能正常运 : 行。分享一下自己 : 的心得,欢迎大牛们拍砖。 : 多google voice主要用于打出,打入的话就都forward到一个SipGate号码就行了。好处: : 1. 多个人用同一个电话时,可以在电话号码上加前缀决定从谁的google voice走。也 : 决定了对方的来电 : 显 : 示显示的是谁的电话。 : 2. 当某人有自己的专属电话时,可以默认所有电话都从他自己的google voice走。 : 假定有一下3个GV 帐号:
| t***n 发帖数: 546 | 5 installed on RT-N16 tomato firmware, asterisk 1.8
max of 2 GV can be forwarded to one sipgate number.
So maybe more sipgate numbers are needed if more than two GV account will be
used.
【在 k***e 的大作中提到】![](/moin_static193/solenoid/img/up.png) : 赞一下! : 你是用lenny还是squeeze?asterisk 1.8吗? 好像多个GV不能forward到同一个 : sipgate吧? : : 处:
| c**s 发帖数: 771 | 6 I guess you can use multiple GV accounts just by using Asterisk, but without Sipgate at all.
Do something like the following:
In Gtalk.conf,
; configure multiple (n) GV gtalks
[GV_account_1]
username=G*********[email protected]
...
context=google_inbound_1
connection=connection_1
...
[GV_account_2]
username=G*********[email protected]
...
context=google_inbound_2
connection=connection_2
...
[GV_account_n]
username=G*********[email protected]
...
context=google_inbound_n
connection=connection_n
...
; end of gtalk.conf
Then in jabber.conf, do the similar
;configure multiple GV connections
[connection_1]
username=G**********[email protected]
secret=(GV_account_1 password)
...
[connection_2]
username=G**********[email protected]
secret=(GV_account_2 password)
...
[connection_n]
username=G**********[email protected]
secret=(GV_account_n password)
...
;end of jabber.com
Finally, in extension.conf
;configure all GV inbound calls.
[google_inbound_1]
exten => G**********[email protected], 1, Dial(SIP/101, 180, D(:1));
...
[google_inbound_2]
exten => G**********[email protected], 1, Dial(SIP/101, 180, D(:1));
...
[google_inbon]
exten => G**********[email protected], 1, Dial(SIP/101, 180, D(:1));
...
Does it work? I don't know. I am having some issues with even only ONE GV account, so I am not going to try anything new. | t***n 发帖数: 546 | 7 The only problem is that in order to receive calls from google voice, you have to uncheck (maybe even have to delete?) all phones from google voice except google chat. Then this google voice account is useless if you want to hook up with your cell, your office .....
That's why for heavily used google voice account, I use sipgate to forwards calls to asterisk.
without Sipgate at all.
【在 c**s 的大作中提到】![](/moin_static193/solenoid/img/up.png) : I guess you can use multiple GV accounts just by using Asterisk, but without Sipgate at all. : Do something like the following: : In Gtalk.conf, : ; configure multiple (n) GV gtalks : [GV_account_1] : username=G*********[email protected] : ... : context=google_inbound_1 : connection=connection_1 : ...
| s**n 发帖数: 449 | 8 no u don't have to.
i have cell / work / gtalk(asterisk) in my google voice.
all three phones will ring.
have to uncheck (maybe even have to delete?) all phones from google voice
except google chat. Then this google voice account is useless if you want to
hook up with your cell, y
forwards calls to asterisk.
【在 t***n 的大作中提到】![](/moin_static193/solenoid/img/up.png) : The only problem is that in order to receive calls from google voice, you have to uncheck (maybe even have to delete?) all phones from google voice except google chat. Then this google voice account is useless if you want to hook up with your cell, your office ..... : That's why for heavily used google voice account, I use sipgate to forwards calls to asterisk. : : without Sipgate at all.
| m**v 发帖数: 373 | 9 请教大牛们几个问题。
1. 我在asterisk 1.8 上配置了gtalk,这样gtalk就一直在线(可以设置away)。 这样
即使我把所有的gmail,gtalk都关掉,别人也看到我在线或者away,这点倒也算了。问
题是如果这个时候别人给我发gtalk信息,我的gmail里面就可以email通知了。本来是离
线的gtalk msg会有email通知的。
2. 多个gtalk都进来没有问题,能不能在电话(我用的是pap2刷的sipura)上用不同的
ringtone来区分不同的gtalk?
3.打出去的时候根据前缀来选择用哪个gtalk,这个还是不太明白,tshen,能否post你
的配置文件参考一下?
4. 如何配置international电话用nonoh或者voipbuster?
谢谢!
have to uncheck (maybe even have to delete?) all phones from google voice
except google chat. Then this google voice account is useless if you want to
hook up with your cell, y
forwards calls to asterisk.
【在 t***n 的大作中提到】![](/moin_static193/solenoid/img/up.png) : The only problem is that in order to receive calls from google voice, you have to uncheck (maybe even have to delete?) all phones from google voice except google chat. Then this google voice account is useless if you want to hook up with your cell, your office ..... : That's why for heavily used google voice account, I use sipgate to forwards calls to asterisk. : : without Sipgate at all.
| t***n 发帖数: 546 | 10 所谓的用前缀来区分,就是本来你要拨打号码xxxxxxxxxx 改为001xxxxxxxxxx 或
002xxxxxxxxxx。 这个
001或002就是前缀,用于区分是哪个用户播出的。
建议先参考http://www.mitbbs.com/clubarticle_t1/voip/31214573_0_1.html, 这里面用到nonoh拨
打中国电话。然后按照我1楼的帖子加上[outbound_User1],[outbound_User2]和[
outbound_Common]
3个context
这样
是离
【在 m**v 的大作中提到】![](/moin_static193/solenoid/img/up.png) : 请教大牛们几个问题。 : 1. 我在asterisk 1.8 上配置了gtalk,这样gtalk就一直在线(可以设置away)。 这样 : 即使我把所有的gmail,gtalk都关掉,别人也看到我在线或者away,这点倒也算了。问 : 题是如果这个时候别人给我发gtalk信息,我的gmail里面就可以email通知了。本来是离 : 线的gtalk msg会有email通知的。 : 2. 多个gtalk都进来没有问题,能不能在电话(我用的是pap2刷的sipura)上用不同的 : ringtone来区分不同的gtalk? : 3.打出去的时候根据前缀来选择用哪个gtalk,这个还是不太明白,tshen,能否post你 : 的配置文件参考一下? : 4. 如何配置international电话用nonoh或者voipbuster?
| | | t***n 发帖数: 546 | 11 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
allow=ulaw
context=google-in
connection=user1
[user2-gtalk]
username=u***[email protected]
disallow=all
allow=ulaw
context=google-in
connection=user2
jabber.conf****************************************
[general]
debug=no
autoprune=yes
autoregister=yes
auth_policy=deny
[dummy]
type=client
serverhost=talk.google.com
username=d***[email protected]
secret=111111
status=xaway
port=5222
usetls=yes
usesasl=yes
statusmessage="This is my Asterisk/Google Voice connection. IMs sent here
will not reach me."
timeout=100
[user3]
type=client
serverhost=talk.google.com
username=u***[email protected]
secret=111111
status=xaway
port=5222
usetls=yes
usesasl=yes
statusmessage="This is my Asterisk/Google Voice connection. IMs sent here
will not reach me."
timeout=100
[user1]
type=client
serverhost=talk.google.com
username=u***[email protected]
secret=111111
status=xaway
port=5222
usetls=yes
usesasl=yes
statusmessage="This is my Asterisk/Google Voice connection. IMs sent here
will not reach me."
timeout=100
[user2]
type=client
serverhost=talk.google.com
username=u***[email protected]
secret=111111
status=xaway
port=5222
usetls=yes
usesasl=yes
statusmessage="This is my Asterisk/Google Voice connection. IMs sent here
will not reach me."
timeout=100
sip.conf*************************************************
[general]
localnet=192.168.1.0/255.255.255.0
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default
is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to
all)
tcpenable=yes ; Enable server for incoming TCP connections
(default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.
0.0 binds to all interfaces)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
nat=yes
allowguest=no
alwaysauthreject=yes
register => 111111:1****[email protected]/111111
register => 111111:1****[email protected]:5060
[SipGate]
nat=yes
type=peer
secret=111111
insecure=invite
username=111111
defaultuser=111111
fromuser=111111
context=sipgate_in
fromdomain=sipgate.com
host=sipgate.com
outboundproxy=proxy.live.sipgate.com
qualify=yes
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
[voipms]
canreinvite=no
context=outbound
host=atlanta.voip.ms
secret=111111 ;your password
type=friend
username=111111 ;your account
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=111111 ;your account
trustrpid=yes
sendrpid=yes
insecure=port,invite
nat=yes ; Uncomment this if your box is behind a NAT
[101]
deny=0.0.0.0/0.0.0.0
permit=192.168.1.0/255.255.255.0
alwaysauthreject=yes
qualify=yes
username=101
secret=111111
type=friend
callerid="Home<101>"
host=dynamic
context=outbound_Common
outgoinglimit=1
incominglimit=1
canreinvite=no
mailbox=101
[102]
;deny=0.0.0.0/0.0.0.0
;permit=0.0.0.0/0.0.0
;permit=192.168.1.0/255.255.255.0
alwaysauthreject=yes
qualify=yes
username=102
secret=111111
type=friend
callerid="User1 IPhone"
host=dynamic
context=outbound_user1
outgoinglimit=1
incominglimit=1
canreinvite=no
mailbox=102
[103]
;deny=0.0.0.0/0.0.0.0
;permit=0.0.0.0/0.0.0
;permit=192.168.1.0/255.255.255.0
alwaysauthreject=yes
qualify=yes
username=103
secret=111111
type=friend
callerid="User2 IPhone"
host=dynamic
context=outbound_user2
outgoinglimit=1
incominglimit=1
canreinvite=no
mailbox=103
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)
GoogleOut=dummy ; Default google voice out
account
[default]
exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten => s,n,Dial(SIP/101,60)
exten => s,n, Hangup
[voicemailboxes]
;main mailbox menu
exten => 100,1,VoicemailMain
exten => 100,2,Hangup
[google-in]
exten => d***[email protected], 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?bridged)
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(
num)})})
exten => _[a-z][email protected], n, Dial(SIP/101, 180, D(:1))
exten => _[a-z][email protected], n(bridged),Bridge(${DB_DELETE(gv_dialout/
channel)}, p)
; No ringing for the following accounts
exten => u***[email protected], 1, Wait, 120
exten => u***[email protected], 1, Wait, 120
; user3 acccount for DISA
exten => u***[email protected],1,Goto(custom-myDISA-auth,1,1)
[sipgate_in]
exten => 111111,1,NoOp(--- ${CALLERID} calling on sipgate (${EXTEN}) ---)
exten => 111111,n,Dial(SIP/101,30)
exten => 111111,n,Answer
exten => 111111,n,Wait,1
exten => 111111,n,Voicemail(u101)
exten => 111111,n,hangup
[custom-myDISA-auth]
exten => _1,1,Wait(2)
exten => _1,n,Answer()
exten => _1,n,Wait(2)
exten => _1,n,SendDTMF(1)
exten => _1,n,Set(TIMEOUT(digit)=15)
exten => _1,n,Set(TIMEOUT(response)=20)
exten => _1,n,Authenticate(111111)
exten => _1,n,Playback(vm-enter-num-to-call)
exten => _1,n,Read(OUTNUM)
exten => _1,n,Goto(outbound_Common,${OUTNUM},1)
exten => _1,n(end),Hangup
[outbound_Common] ;Public phones, need to distinguish ID from the dialing
pattern
exten => _001.,1,Goto(outbound_user1,${EXTEN:3},1)
exten => _002.,1,Goto(outbound_user2,${EXTEN:3},1)
; Begin normal calling using default google voice out, if no ID prefix is
added
exten => _X!,1,Goto(outbound,${EXTEN},1)
exten => _[a-z]!,1,Goto(outbound,${EXTEN},1)
exten => _[A-Z]!,1,Goto(outbound,${EXTEN},1)
[outbound_user1] ; user1's phone, using user1's google voice out
exten => _!,1,Set(GoogleOut=user1)
exten => _!,n,Goto(outbound,${EXTEN},1)
[outbound_user2] ; user2's phone, using user2's google voice out
exten => _!,1,Set(GoogleOut=user2)
exten => _!,n,Goto(outbound,${EXTEN},1)
[outbound]
include => seven-digit
include => custom-international
include => local-devices
;include => tollfree
include => talk-gmail-outbound
include => talk-numeric-outbound
include => dial-uri
[custom-international]
;use SipGate,voipms ....
exten => _01186X.,1,Set(CALLERID(num)=${999999999#})
exten => _01186X.,n,Dial(SIP/${EXTEN}@SipGate,50,trg)
exten => _01186X.,n,Hangup
[local-devices]
exten => _1, 1, Dial(SIP/101,30)
exten => 101, 1, Dial(SIP/101,30)
exten => 102, 1, Dial(SIP/102,30)
exten => 103, 1, Dial(SIP/103,30)
[tollfree]
exten => _411, 1, Dial(SIP/1*********[email protected],60)
exten => _1800NXXXXXX,1,Dial(SIP/${EXTEN}@proxy.ideasip.com,60)
exten => _1888NXXXXXX,1,Dial(SIP/${EXTEN}@proxy.ideasip.com,60)
exten => _1877NXXXXXX,1,Dial(SIP/${EXTEN}@proxy.ideasip.com,60)
exten => _1866NXXXXXX,1,Dial(SIP/${EXTEN}@proxy.ideasip.com,60)
[seven-digit]
exten => _NXXXXXX,1,Set(CALLERID(dnid)=1301${CALLERID(dnid)})
exten => _NXXXXXX,n,Goto(1301${EXTEN},1)
exten => _NXXNXXXXXX,1,Set(CALLERID(dnid)=1${CALLERID(dnid)})
exten => _NXXNXXXXXX,n,Goto(1${EXTEN},1)
[talk-gmail-outbound]
exten => _[a-z][email protected],1,Dial(Gtalk/${GoogleOut}/${EXTEN}@gmail.com)
exten => _[A-Z][email protected],1,Dial(Gtalk/${GoogleOut}/${EXTEN}@gmail.com)
[talk-numeric-outbound]
exten => _1NXXNXXXXXX,1,Dial(Gtalk/${GoogleOut}/${EXTEN}@voice.google.com)
exten => _+1NXXNXXXXXX,1,Dial(Gtalk/${GoogleOut}/${EXTEN}@voice.google.com)
[gv-agi-outbound]
exten => _1NXXNXXXXXX,1,AGI(google-voice-dialout.agi)
exten => _+1NXXNXXXXXX,1,AGI(google-voice-dialout.agi)
[dial-uri]
exten => _[a-z].,1,Dial(SIP/${EXTEN}@${SIPDOMAIN},120,tr)
exten => _[A-Z].,1,Dial(SIP/${EXTEN}@${SIPDOMAIN},120,tr)
exten => _X.,1,Dial(SIP/${EXTEN}@${SIPDOMAIN},120,tr) | m**v 发帖数: 373 | 12 非常非常感谢!按照你的设置, 我的2个GV可以工作了。
【在 t***n 的大作中提到】![](/moin_static193/solenoid/img/up.png) : 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]
| x*g 发帖数: 689 | 13 有人知道怎么设置callcentric? 我按照callcentric网站上asterisk 1.6的guide来设
置,callcentric网站显示registered,asterisk里面也显示registered,打电话进来
总是显示如下错误:
chan_sip.c:21512 handle_request_invite: Sending fake auth rejection for devi
ce ;tag=as49e6d86a
网上查了查也没有找到解决办法
【在 t***n 的大作中提到】![](/moin_static193/solenoid/img/up.png) : 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]
| m**v 发帖数: 373 | 14 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,Segmentation fault,重启asterisk还是同意问
题, 重启dockstar才能解决。。。
【 】 Ctrl-Q 求救 状态 [插入][15,35][ ][ ] 时间【Sat Jun 18 22:40】
【在 t***n 的大作中提到】![](/moin_static193/solenoid/img/up.png) : 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]
| t***n 发帖数: 546 | 15 phonerlite 是一个ATA设备吗?
我又试着给dummy的google voice打了个电话,并且先挂断,并没有出现你说的问题。
exten => d***[email protected], 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?bridged
) 是照前人关于一个GV帐号的配置抄的,具体我也不是很明白。本来下面所有行都应该是
exten => d***[email protected], n*************
但是我改成了
exten => _[a-z][email protected],n*************
不光match d***[email protected],而是match 所有字母开头,@gmail.com结尾的打,也就
是任意gmail帐号。我认为这样的好处是任意配置是否在电话上接听google voice。比
如:
exten => d***[email protected], 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?bridged)
exten => d****[email protected], 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?
bridged)
exten => _[a-z][email protected],n*************
.。。。。。。。。。。。。。。。。。。。
这样dummy和dummy2的来电会振铃。反之
exten => d***[email protected], 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?bridged)
exten => _[a-z][email protected],n*************
.。。。。。。。。。。。。。。。。。。。
exten => d****[email protected], 1, Wait, 120
则dummy2的来电不会振铃
你这个死机还真不知道怎么回事。按说无论配置有什么问题,顶多拨打出问题,不会死
机退出的。
d
bridged
channe
【在 m**v 的大作中提到】![](/moin_static193/solenoid/img/up.png) : 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
| m**v 发帖数: 373 | 16 phonerlite是个免费softphone。
我发现那个segmentation fault只是在run asteriks -cvvvvv而且是对方先挂电话的时
候,如果asterisk运行在后台就不会segmentation fault,但是这个warning还是在log
file里面:
features.c: Bridge failed because channel does not exists or we cannot get
its lock
bridged
该是
【在 t***n 的大作中提到】![](/moin_static193/solenoid/img/up.png) : phonerlite 是一个ATA设备吗? : 我又试着给dummy的google voice打了个电话,并且先挂断,并没有出现你说的问题。 : exten => d***[email protected], 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?bridged : ) 是照前人关于一个GV帐号的配置抄的,具体我也不是很明白。本来下面所有行都应该是 : exten => d***[email protected], n************* : 但是我改成了 : exten => _[a-z][email protected],n************* : 不光match d***[email protected],而是match 所有字母开头,@gmail.com结尾的打,也就 : 是任意gmail帐号。我认为这样的好处是任意配置是否在电话上接听google voice。比 : 如:
| t***n 发帖数: 546 | 17 你能把debug信息贴出来吗?从拨电话开始
log
get
【在 m**v 的大作中提到】![](/moin_static193/solenoid/img/up.png) : phonerlite是个免费softphone。 : 我发现那个segmentation fault只是在run asteriks -cvvvvv而且是对方先挂电话的时 : 候,如果asterisk运行在后台就不会segmentation fault,但是这个warning还是在log : file里面: : features.c: Bridge failed because channel does not exists or we cannot get : its lock : : bridged : 该是
| m**v 发帖数: 373 | 18 这里是一个完整的call的信息,对方挂电话就直接segmentation fault了。log里面没有
任何信息。
*CLI> -- Executing [D***[email protected]@google-in:1] GotoIf("Gtalk/+XXX-152d
", "0?bridged") in new stack
-- Executing [D***[email protected]@google-in:2] NoOp("Gtalk/+XXX-152d", "Call
erid +**[email protected]/srvres-MTAuMjIwLjIwNC4yMTo5ODQw") in new stack
-- Executing [D***[email protected]@google-in:3] Set("Gtalk/+XXX-152d", "CALLE
RID(num)=+XXX") in new stack
-- Executing [D***[email protected]@google-in:4] Set("Gtalk/+XXX-152d", "CALLE
RID(name)=") in new stack
-- Executing [D***[email protected]@google-in:5] Dial("Gtalk/+XXX-152d", "SIP/
101, 180, aD(:1)") in new stack
== Using SIP RTP CoS mark 5
-- Called 101
-- SIP/101-00000000 is ringing
-- SIP/101-00000000 answered Gtalk/+XXX-152d
-- Sending DTMF '1' to the calling party.
Segmentation fault
debian:~/bin#
以下是我在phonerlite里面先挂电话的信息,最后一个warning也在log文件里。
*CLI> -- Executing [D***[email protected]@google-in:1] GotoIf("Gtalk/+XXX-c53e
", "0?bridged") in new stack
-- Executing [D***[email protected]@google-in:2] NoOp("Gtalk/+XXX-c53e", "Call
erid +**[email protected]/srvres-MTAuMTIuMTYzLjM3Ojk4MjI=") in new stack
-- Executing [D***[email protected]@google-in:3] Set("Gtalk/+XXX-c53e", "CALLE
RID(num)=+XXX") in new stack
-- Executing [D***[email protected]@google-in:4] Set("Gtalk/+XXX-c53e", "CALLE
RID(name)=") in new stack
-- Executing [D***[email protected]@google-in:5] Dial("Gtalk/+XXX-c53e", "SIP/
101, 180, aD(:1)") in new stack
== Using SIP RTP CoS mark 5
-- Called 101
-- SIP/101-00000001 is ringing
-- SIP/101-00000001 answered Gtalk/+XXX-c53e
-- Sending DTMF '1' to the calling party.
== Spawn extension (google-in, D***[email protected], 5) exited non-zero on 'Gta
lk/+XXX-c53e'
[Jun 19 11:09:15] WARNING[12675]: chan_sip.c:3551 retrans_pkt: Retransmissio
n timeout reached on transmission 1******************************[email protected]
.140:5060 for seqno 103 (Critical Response) -- See https://wiki.asterisk.org
/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response
【在 t***n 的大作中提到】![](/moin_static193/solenoid/img/up.png) : 你能把debug信息贴出来吗?从拨电话开始 : : log : get
| s**n 发帖数: 449 | 19 thanks. can u please explain what is user3/myDISA for?
【在 t***n 的大作中提到】![](/moin_static193/solenoid/img/up.png) : 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]
| t***n 发帖数: 546 | 20 不好意思,搞不定你这种疑难杂症。
没有
152d
Call
CALLE
CALLE
【在 m**v 的大作中提到】![](/moin_static193/solenoid/img/up.png) : 这里是一个完整的call的信息,对方挂电话就直接segmentation fault了。log里面没有 : 任何信息。 : *CLI> -- Executing [D***[email protected]@google-in:1] GotoIf("Gtalk/+XXX-152d : ", "0?bridged") in new stack : -- Executing [D***[email protected]@google-in:2] NoOp("Gtalk/+XXX-152d", "Call : erid +**[email protected]/srvres-MTAuMjIwLjIwNC4yMTo5ODQw") in new stack : -- Executing [D***[email protected]@google-in:3] Set("Gtalk/+XXX-152d", "CALLE : RID(num)=+XXX") in new stack : -- Executing [D***[email protected]@google-in:4] Set("Gtalk/+XXX-152d", "CALLE : RID(name)=") in new stack
| | | t***n 发帖数: 546 | 21 这个是这样操作的。
1. 拨打user3的电话号码
2.输入密码
3.输入要拨打的号码,就像使用家里的电话一样。
好处:
1. 出门在外要打国际长途,可以把user3的号码当local access。
2. 可以在外面通过分机号拨打家里的某台电话(这台电话不能被直接访问,以防打扰
)。
其他的用处我也没有想到,欢迎大家挖掘一下
【在 s**n 的大作中提到】![](/moin_static193/solenoid/img/up.png) : thanks. can u please explain what is user3/myDISA for?
| x*g 发帖数: 689 | 22 我按葫芦画瓢也弄了一个, 中间总是直接退出了,好像少load了一些module,autoloa
d设置成yes也不行。输入密码正确以后说让我输入要打的电话以后就显示没有read这个
application,马上断线了。你的asteriskload了哪些module?
-- Playing 'agent-pass.gsm' (language 'en')
-- Playing 'auth-thankyou.gsm' (language 'en')
-- Executing [1@custom-my-auth:8] Playback("Gtalk/+xxx-f4af", "vm-enter-
num-to-call") in new stack
-- Playing 'vm-enter-num-to-call.gsm' (language 'en')
[Jun 19 20:42:53] WARNING[13055]: pbx.c:4054 pbx_extension_helper: No applic
ation 'Read' for extension (custom-my-auth, 1, 9)
== Spawn extension (custom-my-auth, 1, 9) exited non-zero on 'Gtalk/+xxx-f
4af'
【在 t***n 的大作中提到】![](/moin_static193/solenoid/img/up.png) : 这个是这样操作的。 : 1. 拨打user3的电话号码 : 2.输入密码 : 3.输入要拨打的号码,就像使用家里的电话一样。 : 好处: : 1. 出门在外要打国际长途,可以把user3的号码当local access。 : 2. 可以在外面通过分机号拨打家里的某台电话(这台电话不能被直接访问,以防打扰 : )。 : 其他的用处我也没有想到,欢迎大家挖掘一下
| t***n 发帖数: 546 | 23 需要
load => app_read.so
autoloa
enter-
applic
【在 x*g 的大作中提到】![](/moin_static193/solenoid/img/up.png) : 我按葫芦画瓢也弄了一个, 中间总是直接退出了,好像少load了一些module,autoloa : d设置成yes也不行。输入密码正确以后说让我输入要打的电话以后就显示没有read这个 : application,马上断线了。你的asteriskload了哪些module? : -- Playing 'agent-pass.gsm' (language 'en') : -- Playing 'auth-thankyou.gsm' (language 'en') : -- Executing [1@custom-my-auth:8] Playback("Gtalk/+xxx-f4af", "vm-enter- : num-to-call") in new stack : -- Playing 'vm-enter-num-to-call.gsm' (language 'en') : [Jun 19 20:42:53] WARNING[13055]: pbx.c:4054 pbx_extension_helper: No applic : ation 'Read' for extension (custom-my-auth, 1, 9)
| x*g 发帖数: 689 | 24 加了app_read可以了, 谢谢。 奇怪的是autoload=yes也不行。
打中国用哪个可以显示callerid?
【在 t***n 的大作中提到】![](/moin_static193/solenoid/img/up.png) : 需要 : load => app_read.so : : autoloa : enter- : applic
| s**n 发帖数: 449 | 25 thanks.
【在 t***n 的大作中提到】![](/moin_static193/solenoid/img/up.png) : 这个是这样操作的。 : 1. 拨打user3的电话号码 : 2.输入密码 : 3.输入要拨打的号码,就像使用家里的电话一样。 : 好处: : 1. 出门在外要打国际长途,可以把user3的号码当local access。 : 2. 可以在外面通过分机号拨打家里的某台电话(这台电话不能被直接访问,以防打扰 : )。 : 其他的用处我也没有想到,欢迎大家挖掘一下
| x*g 发帖数: 689 | 26 我也是按照你的做了一个这样的,一直好好的,今天突然不行了,打user3的电话
asterisk就有如下的错误:
WARNING[3025] file.c: Failed to write frame
重启asterisk有时就可以了,但是打几次又出现上面的错误就不行了。
上网google了一会儿也找不到怎么回事,大家给看看?
【在 t***n 的大作中提到】![](/moin_static193/solenoid/img/up.png) : 这个是这样操作的。 : 1. 拨打user3的电话号码 : 2.输入密码 : 3.输入要拨打的号码,就像使用家里的电话一样。 : 好处: : 1. 出门在外要打国际长途,可以把user3的号码当local access。 : 2. 可以在外面通过分机号拨打家里的某台电话(这台电话不能被直接访问,以防打扰 : )。 : 其他的用处我也没有想到,欢迎大家挖掘一下
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