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Provide basic FreeSWITCH configuration

[updated job: no bare minimum requirement, no source code build]

I need a FreeSWITCH (FS) configuration for the following functionality:

1. Interconnect internal endpoints

2. Link with an existing Asterisk PBX

3. Link with an external SIP trunk for incoming and outgoing calls

4. Configure basic CDR, voicemail, IVR and conference

5. Configure presence/BNL

6. Configure chat on Linphone endpoints (optional)

Details

1. Interconnect:

* 2 Linphone endpoints, extensions 100 and 101

* 1 Linphone endpoint and 1 Yealink T41S phone on the same extension 102 (both should ring simultaneously on incoming call)

2. Link with an existing internal Asterisk PBX that has 2 more endpoints with extensions 200 and 201. They should be accessible to the FS (FreeSWITCH) endpoints and the FS extensions (100-102) should be accessible to the Asterisk endpoints.

3. Link with an external SIP trunk provider for incoming and outgoing calls.

For outgoing calls, all numbers dialed on internal endpoints that match full international number format should go via the external SIP trunk, all numbers that are longer than 3 digits (i.e. that are not internal extensions) but don't match the international format should be rejected as invalid with a voice message (e.g. "the number you dialed is incorrect"). The FS extension 101 should NOT be able to call outside (only internal calls to other extensions).

For incoming calls from the external SIP trunk, there are 2 external (PSTN) numbers (e.g. +1 905 878 5000 and +1 905 878 5001) - when receiving a call for the first number, it should be redirected to the ext. 100 at FS, for the second number to the ext. 200 at Asterisk (i.e. +1 905 878 5001 coming from external SIP trunk → FS → Asterisk → ext. 200).

Endpoints connected to Asterisk should be able to make outside calls too via the external SIP trunk (e.g. ext. 200 → Asterisk → FS → external SIP trunk).

4. Configure basic CDR, voicemail, IVR and conference

* Basic 2-leg CDR to MySQL with failover to CSV files;

* Basic voicemail for specific extensions (e.g. 100 and 101 after no answer for 30s), only the configured extensions should be able to access their voicemails (with changeable password);

* Basic IVR (e.g. on incoming call from the external SIP trunk): play a welcome message, then accept entering an extension from a specific set (e.g. 100-102) with a timeout for automatic redirect to a predefined extension (e.g. 100). A specific endpoint (e.g. 102) should be able to record/change the welcome message entering a password.

* Basic conference: admin and normal users call an extension to join (both with passwords); on joining the conference FS should ask for a user prompt, then announce the user to the conference; possibility for the admin to invite (dial) users from the conference.

5. Configure presence/BLF: presence on Linphone endpoints (available, busy, away, etc.) and BLF on Yealink phone for the extensions connected directly to FS (100-102).

6. Configure chat on Linphone endpoints (with SIP SIMPLE) - optional, please specify if you are sure you'll be able to do it or not.

Additional details:

There is NO need for NAT traversal. The FS itself, its endpoints and Asterisk are on the same private network (say 10.0.1.0/24); the external SIP trunk has a public static IP (say 180.180.80.1), the internal network also has a public static IP (say 190.190.120.8). The required ports could be opened as needed.

The communication should always go via FS (no direct RTP). The endpoints configured with FS should be using Opus/[url removed, login to view] and VP8/[url removed, login to view] codecs. When communicating with the external SIP trunk provider and Asterisk, should be using [url removed, login to view]

Everything should be provided as basic instructions and configuration files (e.g. the entire FS conf directory, instructions for Linphone and Yealink basic settings, etc.). It is NOT needed to deploy the configuration on my server, I'll do it myself with your instructions/configs.

Skills: Asterisk PBX, FreeSwitch, VoIP

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About the Employer:
( 0 reviews ) Buenos Aires, Brazil

Project ID: #14698403

7 freelancers are bidding on average $257 for this job

tektrix1

I m a voip professional having extensive experience of 15+ years in voip/asterisk/freeswitch applications. I have delivered many projects based on asterisk and freeswitch contact centers and IVRs , OpenSips and Asteris More

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waheni

extensive experience with voip Relevant Skills and Experience linux ,asterisk ,voip Proposed Milestones $194 USD - first $194 USD - final

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hstoddart

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optools87

Our Many clients are using Free PBX inbound and outbound system for calling in UAE and KWI Relevant Skills and Experience 11 year Experience in asterisk ,vicidial,goautodial and Free PBX Proposed Milestones $77.5 USD More

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nhamid6

Hello I am an experienced developer that has over 7 years of practice in FreeSwitch. I may be able to help you. Please get in touch with me. Stay tuned, I'm still working on this proposal.

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