Asterisk 1.8 新功能真的幾方便好用

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2^#

發表於 2011-1-4 19:06 | 只看該作者

For google voice, the direct dial through gizmo5+sipsorcery(cannot register for free now) has been used for a while.

Next week, when the IP01 is arrived. The migration to IP01 beginz. The atom-PC might used for other objective, like skype trunk using siptosis.

bubblestar

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3^#

發表於 2011-1-4 19:41 | 只看該作者

I could not use direct dial as I have been using Sipgate instead of Gizmo5, hence, not aware of this changes. BTW and for you info, Asterisk 1.8 no longer supports dialing from its Google Web. Despite of that, Asterisk 1.4 can still use this traditional method.

Speaking of Skype trunk, this can be a good topic for me to learn. If it rings a bell to those who are familiar to this, please give us a hand.

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4^#

發表於 2011-1-4 20:01 | 只看該作者

回復 3# bubblestar

Try the siptosis, it worths the money spent

http://www.mhspot.com/sts/siptosis_skype_trunk_howto.html

In an ATOM PC, you can easily get 4 channels working

bubblestar

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5^#

發表於 2011-1-4 20:40 | 只看該作者

Sounds interesting! Thanks.

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6^#

發表於 2011-1-4 21:04 | 只看該作者

回復 5# bubblestar

Just US$23 you can get the program and trunk builder. Mine is very old. Just US$9.99 for both

Qnewbie

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7^#

發表於 2011-1-4 21:25 | 只看該作者

For the free siptosis, installed in Ubuntu(not atom PC), it responses a bit slow. Perhaps the pay-version would be better.

A lot of callback service with Skype as backbone can be found in taobao. Might I make a query to ckleea C-hing, could pay-version siptosis be used for this end?

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8^#

發表於 2011-1-4 21:36 | 只看該作者

回復 7# Qnewbie

I am not sure what you mean. For my setup, the quality is just the same as usual VOIP or PSTN line.
If you look at this for the changes, it just worths the money spent

http://www.mhspot.com/stsforum/viewtopic.php?id=316

Mine is still 2009 Oct one

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9^#

發表於 2011-1-4 22:10 | 只看該作者

剛用google voice互打，網絡起碼經兩次太平洋，同固網基本上一樣。

Qnewbie

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10^#

發表於 2011-1-4 22:12 | 只看該作者

Oh, the siptosis closes the skype connection 5 to 10 seconds after I hang up. That is why I said slow response.

The skype callback used by taobao seller after google searching is called QQ callback(?). The detailed info is unknown. But if a single account is involved, it should be some kind of conference to connect two PSTN users(+skype acc. self). The question might be formulated by following: could we use a single skype account to call two PSTN user and connect them together with conference with siptosis?

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11^#

發表於 2011-1-4 22:21 | 只看該作者

回復 10# Qnewbie

I believe so. Just like conference in asterisk

Skype 1 -> skype 2 connected by siptosis to asterisk -> DTMF into conference room
PSTN user -> Asterisk box -> conference room
Skype 3 -> skyper 4 connected by siptosis to asterisk -> conference room
VOIP user -> Asterisk box -> conference room

I have not tested this before. But as far as I know it should work