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c*********i
发帖数: 2749 | 3 ［欢迎版主加入宝典有关GV和自由泡的文章中，谢谢！］
如果直接使用自由泡客户端接听GV的验证来电，无法通过验证，前两天看到大熊版主提
示，可以通过telephone tone generator通过验证，于是觉得应该不需要去找一台电话
机，原理是通过不同数字键发出的音频不同。于是今天搜了一下，发现谷歌市场就有这
样一个app：
DTMF Tone Generator
链接：https://play.google.com/store/apps/details?id=com.amknott.ToneGen
验证方法，先将这个tone发声app安装在另外一台安卓手机，然后接听GV验证来电，然
后这个app 按数字即可，记得打开扬声器，让tone发声app的声音进入接收验证来电的
那台手机的麦克风。
终于，可以通过自由泡的app，在大陆免翻墙接听来自GV转发的电话和短信了。 |
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P*****s
发帖数: 375 | 5 atuma您实在是太牛了！！！
用您的第一招，借用了个安卓手机直接就把GV验证搞定了。陈同学用两台安卓手机的方
案加DTMF的方法貌似不需要。
苹果手机不能验证GV来电码。安卓却可以！
=====================
另外，有data可以用GV号码就可以打进打出的话，自由泡漏接电话不重要了吧？
因为只是用自由泡的data。GV号码是联系外界的主号，我的理解对吗?
谢谢！ |
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l*****0
发帖数: 320 | 6 g729的问题是不抗丢包&有各种兼容问题，尤其是DTMF
最后还是用回g711了 |
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H***a
发帖数: 189 | 7 Verify by call是个怎么不work? Phone won't ring or call disconnect or no
audio or DTMF not recognized ? |
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l*******s
发帖数: 7316 | 8 怀疑是按键声音不对。弄个大声音的有线电话在旁边按应该能通过。
：Verify by call是个怎么不work? Phone won't ring or call disconnect or
no
：audio or DTMF not recognized ? |
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H***a
发帖数: 189 | 9 DTMF problem is a typical Issue for some VoIP providers if they don't do it
right.
你说的没错，弄个声音大的电话放在听筒那里试试。有可能work. |
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w*******y
发帖数: 60932 | 10 Panasonic KX-TG9333T DECT 6.0 Expandable Digital Cordless Phone with All-
Digital Answering System, Call Block, Night Mode
Price: $39.99
Mfr: Panasonic
Model: KX-TG9333T
FRYS.com #5515340
UPC: 037988479481
Detailed Description
(Manufacturer # KX-TG9333T )
Link:
http://www.frys.com/product/5515340
Includes 1x Handset with Base, 2 Handsets with charger
Up to 6 handsets capability
60 Channels
Frequency: 1.9 GHz
Battery: Ni-MH (AAA x 6)
Battery Life: 5 hours talk time, 11 days... 阅读全帖 |
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r****t
发帖数: 10904 | 11 我现在才发现我的 asterisk + pap2 + GV 打信用卡公司输入卡号时候不能被 recognize.
以前读过这个贴，所以 asterisk 里面设了 inband 了：
...
Subscriptions: Yes
Overlap dial : No
DTMFmode : inband
Timer T1 : 500
Timer B : 32000
...
为啥还是没法打呢？ |
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a9
发帖数: 21638 | 12 pap2里也改了试试看？
recognize. |
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r****t
发帖数: 10904 | 13 a9 老大威武，这个 pap2 里面改了立即就 work 了！谢谢！ |
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k****t
发帖数: 2288 | 15 设置你的ATA，用inband吧！
另外codec用729u。
我原来是通过asterisk来接google voice，也是碰到同样的问题，后来我就全部用
inband DTMF，codec用729u来解决这个问题。 |
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i**w
发帖数: 883 | 16 Asterisk装在router上，性能跟不上吧。还是放在NAS上比较好，24*7开机，还可以玩
玩fancy的功能，例如DTMF, call back, morning call等等 |
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i**w
发帖数: 883 | 17 Asterisk装在router上，性能跟不上吧。还是放在NAS上比较好，24*7开机，还可以玩
玩fancy的功能，例如DTMF, call back, morning call等等 |
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k****t
发帖数: 2288 | 18 我前面有帖子讲这个问题的。
主要就是设置DTMF的问题！ |
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e**u
发帖数: 409 | 19 你要写dialplan的时候中间wait几秒，然后再发送DTMF就应该可以 |
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a9
发帖数: 21638 | 20 折腾了一天，现在和gtalk语音没问题（本来就是正常的）
和gmail里的网页语音聊天没问题（原来不支持）
能打电话（原来打不出去）
不能接电话（发送dtmf有问题，因为接google voice来的电话时，提示按1通话，按2留
言) |
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w****a
发帖数: 6326 | 21 有什么voip可打 800 的免费的，而且dtmf support比较好的。 |
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a9
发帖数: 21638 | 22 dtmf mode 用inbound
pinless。 |
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p**e
发帖数: 533 | 23 是这个吗？DTMF Tx Method:？
我在我的ATA里只看到
Inband
AVT
INFO |
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f*u
发帖数: 5576 | 26 debug asterisk: asterisk -rvvvv
make call to your gvoice to see DTMF(1) sent out or not |
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C*******1
发帖数: 422 | 27 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,) |
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f*u
发帖数: 5576 | 30 debug asterisk: asterisk -rvvvv
make call to your gvoice to see DTMF(1) sent out or not |
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C*******1
发帖数: 422 | 31 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,) |
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r*****8
发帖数: 2697 | 32 谢谢!
不过在我的神座上还缺一个module: app_senddtmf.so
还有你的res_rtp_asterisk名字不全, 应该是res_rtp_asterisk.so
下面是我的更新:
[modules]
autoload=no
load => format_pcm.so ; uLaw/ALaw
load => codec_ulaw.so ; mu-Law
load => format_g726.so ; Raw G.726
load => codec_g726.so ; g-726
load => format_gsm.so ; Raw gsm
load => codec_gsm.so ; gsm Coder/Decoder
load => app_dial.so ; Dialing
load => app_macro.so ; Extension Macros
load => app_playback.so ... 阅读全帖 |
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F******k
发帖数: 7375 | 33 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 |
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s**n
发帖数: 449 | 34 thanks for the info. currently i have asterisk1.8+gv setup.
however, sometime i have to press 1 to answer incoming gv call.
i have already disabled call screening in gv setting
and tried to send DTMF in asterisk. but it is not working consistently.
do u know if obi has this problem? |
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a9
发帖数: 21638 | 35 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 |
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t***y
发帖数: 741 | 36 supposedly, it's developed by the same guys behind the PAP2
However, I had problems with obi as well. I totaly have 3 units. One of them
DOA and I had to get a replacement. One of them malfunctioned (phone port
dead) after two months, then I had to get it replaced directly from obi (as
it was out of amazon's 30 day window).
Also, it had some problems with DTMF. Instead, my three PAP2T are working like wonder. The reason I used obi are mainly remotely admin. I did not care about GV as voip rate i... 阅读全帖 |
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c**s
发帖数: 771 | 37 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. |
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d********g
发帖数: 10550 | 39 这个在asterisk里怎么设，假如客户端不可以改的话？比如有些softphone是没有选项
的，连asterisk server上能看到DTMF输入，但是GV不认。ATA的客户端倒是可以改这个 |
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l***h
发帖数: 9308 | 40 我也是看别人说改这个
SPA系列里面基本上DTMF都要改成inband |
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d********g
发帖数: 10550 | 41 在*上把dtmfmode设成rfc2833或者inband好像都不行（sip.conf）。可以打出，但接听
键按了没有反应
Using SIP RTP CoS mark 5
-- Called SIP/danielfeng
-- SIP/danielfeng-00000000 is ringing
-- SIP/danielfeng-00000000 is ringing
-- SIP/danielfeng-00000000 answered Gtalk/+1XXXXXXXXXX-XXXX
-- Sending DTMF '1' to the calling party.
ATA客户端里设没有问题，softphone不能设 |
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