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FreeSwitch PBX Expert Needed

I'm looking to outsource a FreeSwitch PBX expert to help me update my existing server configuration (Ubuntu based).

First off, I need to add a gateway and a couple more dongles to it AND extract a particular variable from the SIP header that the new gateway supplies. The variable needs to be accessible in dialplan in expressions like this one: <action application="set" data="api_hangup_hook=jsrun [url removed, login to view] ${the_variable_name}"/>.

Please let me know how you can help me with this.

There's more work for the successful candidate. Apply!

Header sample:

INVITE sip:456ХХХХХХХ[url removed, login to view] SIP/2.0

Via: SIP/2.0/UDP [url removed, login to view];branch=z9hG4bK2cd64e28

Max-Forwards: 70

From: “123XXXXXXX”<sip:>;tag=as773aa379

To: <sip:456XXXXXXX(a)[url removed, login to view]>

Contact: <sip::5060>

Call-ID: 0a0af71710dc99586a598cff741c203f

CSeq: 102 INVITE

User-Agent: Limatelecom Trunk CPE

Date: Tue, 11 Jul 2014 12:18:35 GMT

Diversion: “12329084XX”<sip:>;reason=unconditional

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 253

Навыки: Asterisk PBX, Телекоммуникационная техника, VoIP

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О работодателе:
( 13 отзыв(-а, -ов) ) Sevastopol, Russian Federation

ID проекта: #6257405

1 фрилансер подал заявку на эту работу; средняя заявка - $277

ricky1618

4 years experience in Asterisk, Elastix, Cisco, Python, PHP Programer. I'm the general manager in a VoIP integrator of my country with a great reference.

$277 USD за 2 дней(-я)
(0 отзывов(-а))
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