[OpenSIPS-Users] Anybody is interested to develop this scenario using OpenSIPS?

[Satish Patel]

I got confused in your diagram so i just wanted to clean, anyway i think
your diagram should be like following


[UA]-------------[Opensips]--------------[SIP Provider]
                     |           |
                     |           |
                     |           |
                    [Asterisk]
                    [Asterisk]


This is what i understand, Please correct me if i am wrong:

1. Asterisk with handle VoiceMail, Conference, IVR etc service.
2. Opensips will send all Outbound calls to PSTN directly.
3. Did you create Asterisk Database for Username etc? I think we may need
to Asterisk view table to both can read user DB. Its not a big deal.


Regarding hour it may it may take 5/6 hours or may be less or more depend
on what kind of issue we see. I will charge overall. 15,000/-

Let me know if i am missing anything.




On Fri, Apr 17, 2015 at 2:10 AM, Chandramouli P <mouli123 at gmail.com> wrote:

> Hello All,
>
> If anybody is interested to develop this below written scenario using
> OpenSIPs, please let me know.
>
> Global SIP Users ---> OpenSIPS ---> Asterisk media server1 -----------
>
> |                                                  |---------- VoIP
> provider for PSTN calls
>
> |                                                  |
>                                                 ---> Asterisk media
> server2 -----------
>
> *Assumptions:*
> OpenSIPS public IP address (eth0): 104.131.65.66
> OpenSIPS private IP address (eth1): 10.10.10.1
> Asterisk media server1 private IP address (eth1): 10.10.10.2
> Asterisk media server2 private IP address (eth1): 10.10.10.3
> MySQL DB server private IP address (eth1): 10.10.10.4
> VoIP provider public ip address: 123.456.789.111
>
> 1) All servers are hosted in Digital ocean and in private network
> 2) All SIP users, voice mail users, dial rules will be stores in MySQL
> database
> 3) I must give OpenSIPS proxy server public ip address in to my VoIP
> provider. My provider will allow incoming/out going traffic through this IP
> address only. But, call should go through our media servers only. Because,
> dial rules will be stored in MySQL database.
> 4) SIP users will connects to openSIPs proxy server from globally
> 5) I will provide you the whole environment with the installed OpenSIPs
> (Ubuntu), installed Asterisk (CentOS) servers, and installed MySQL database
> tables.
> 6) I will configure Asterisk in real time and data base.
>
> *Task:* You need to provide me OpenSIPs working configuration file to
> fulfill the below needs for the above environment:
> 1) Nat traversal
> 2) SIP registrations through proxy (As I said, we store all sip users
> details in MySQL database table)
> 3) Load balancing (We will give two media servers) with fail over
> 4) PSTN inbound/outbound calling through media servers by using MySQL data
> base tables (Because, we store users dial rules in db table). But, we give
> our Proxy server ip address to our VoIP provider for authentication purpose.
>
> Please do not reply me, if you are a learner. Only experienced
> professional with OpenSIPS are welcome.
>
> Thank you.
> Chandra.
>
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>
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