VoIP, openSER - Asterisk

This project received 18 bids from talented freelancers with an average bid price of $1306 USD.

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I am looking for an experienced Asterisk/openSER programmer to undertake development in my VoIP-portal. This allows users to sign-up, register their phone, make calls with other registered users and to PSTN. Site IP: [url removed, login to view]

I have SER/Asterisk/AstCC/MySQL/PHP components but SER/RTPproxy doesn't work well and I need a new openSER installation and configuration for services listed below.
This bidding is for the installation/configuration from all softwares you are needed incl. additional scripts. You should make the adaptation in my existing website and if you won't finish the whole project, you should provide my PHP programmer with specifications and information.

Only coder with previous experience on openSER/Asterisk are accepted.
70% of the payment will be given once the job is completed and working.
You will have to give 30 days free support for testing and removing any bugs on the site after the website is ready. The remaining 30% will be given to you after the testing days.

If you do not agree on the terms mentioned above, please DON'T waste your time and DON'T bid. If you succeed completing this project, more features will be awarded to you soon after.

Please tell me what is your method? You can install openSER and configure Asterisk with the needed services or reinstall the whole VoIP server part. My choice will be on the basis of your experience not just on the price. Finally a need a stable VoIP system.

If you are a good PHP coder (smarty) you can develop my webinterface too. This job contains filds:
- Development of userinterface and admininterface, possibility to edit the services listed below
- Implementation of additional payment methods (paypal, moneybookers etc)
- Optimizations for SEO (mod rewrite urls, css, valid HTML)

This will be paid by arrangement. In this case please make a offer.

Services needed:

* Internal calling
* PSTN calling/ billing
* DID forwarding
* Call forwarding
* Call forwarding on busy
* Call forwarding on no answer
* Voicemail(Using feature server Asterisk as voicemail server)
- I need just a simple configuration, with new text, and voice file from me (translation). PIN request.
* LCR(Least Cost Routing)
* Enum Support
* Video calling
* NAT traversal support (RTPproxy, Madisproxy or same like SBC)
This is the pretentious and sophisticated part on the configuration. Please bid if you are a NAT traversal expert. Please tell me what kind of solution you want to use.
* Text Messaging
- should be able to use openWengo
- Jabber XMPP Voice, IM support
- Webphone like A2billing (user interface)
* Callthrough
* Click to Call (Callback)
1 Customer enters source and destination numbers in Click to Call web page.
2 System calls source number
3 When user picks up the phone, he is connected to destination number
* Web Call Me
A customized button placed in a website can offer customers to get an instant connection with live sales or support persons.
1. User clicks web "Call Me" button on website.
2. System calls user.
3. When user picks up the phone, system connects the user with sales/support person.
* Ability to activate/deactivate service at any time

Additional tools needed:

* Calling Card script (You can use your favorit Calling card sollution, or AstCC)
* Call Statstic

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