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siptosis - killer sip and skype integration

It is unfortunate that there will be no more support for non paid version or commerical version for purchase.
In the last few weeks, I have been playing with the siptosis trunk setup. Now it is really fantastic.
Multiple trunks
skype users based call in rule
outgoing to different skype users by speed dial
watchdog to check config changes and reload

Yes, you are right.  Different Skype clients have their own ways to access the keypad to send DTMF for further dialing.  Nokia E-series and iPhone, for examples, are quite different but they can use the hidden keypads without problem.

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Update on the use of mobile skype client.

With the configuration of siptosis, the way of handling incoming calls is very flexible.

We can go directly to an asterisk extension, call forward, call back, ivr, etc

Very often , we need to find the keypads and send the DTMF. In skype, i think they won't expect we do DTMF very often and so the keypad is often hidden or not existing. We need to find out in every skype clients we use.

Once you get it, you can do whatever you want in high chance of success.

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回復 7# bubblestar
You are in the right track. You can change the other files as well so that each channel has its own config.

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回復 5# bubblestar


[Solved]

I have tweaked the related stsTrunk_xx.cfg files and divert incoming calls to different ultimate skype users in my server.  The key is to identify which SkypeToSipAuth.props that the incoming call should go.  Now, I split this file into:

SkypeToSipAuth_01.props and SkypeToSipAuth_02.props in which they have different destination settings.

Anyway, I enjoy the learning process in this siptosis multi channels installation.

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回復 5# bubblestar

Not sure what you mean.

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I have successfully swap from single channel to multi channels in siptosis last night.  All outgoing call tests are great.  Incoming calls are also of no problem basically.

Does anyone know how to setup incoming calls to go into any desired skype account?  I ask as all my existing calls from outside now only go into the same skype account, ie. stsTrunk_01).

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I tried last night with bublestar again

Skype from Japan -> siptosis-> asterisk-> sip call to pstn

Excellent quality, no delay, talked for more than 15 min.

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If you can add a remote skype/PSTN phone, you can almost connect to every system you like with guarantee quality

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If you wish, one can install many skype instances within a linux server. This gives one great flexibility in incoming and outgoing calls.

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