Possible to use the SIP/Voip from voice on FTTH with Freepbx or something similar?

@shashankb Okay, so not changing the current WAN configuration these are the steps that I followed:
  1. I went to PON WAN page to select new link. There, from the drop down, I clicked new link.
  2. Enabled VLAN, set it to 1831, 802.1p_Mark was left null
  3. Changed channel mode to IPoE
  4. Changed connection type to VOICE_INTERNET
  5. WAN settings was changed to DHCP and DNS request was also enabled
  6. None of the ports were mapped
  7. Clicked on apply changes, but the connection never went up.
Also, on an unrelated note, when I tried pinging kl.ftthvoip.bsnl.in(10.187.6.12) that also resulted in failure.
Is there any condition that either VOICE or only VOICE_INTERNET exists, but not both? I need voice for my land line, but SIP is good to have at my home.
 
You cannot ping the SIP server through the Internet you are using . Internet and Voice both are two different networks in simple language...You will need a router that runs Openwrt so that you can get to the SIP server by putting the Voice in bridge mode
 
You can add dialplan ( replace +91 country code, append 0 before mobile numbers) on siproxd config itself. No need to configure any sip client. Here is what I had done with my openwrt router.

First, install siproxd-mod-regex

opkg install siproxd-mod-regex

Then these commands

Code:
uci add_list siproxd.general.load_plugin 'plugin_regex.so'

uci add_list siproxd.general.plugin_regex_desc='country_code'
uci add_list siproxd.general.plugin_regex_pattern='^(sips?:)\+91'
uci add_list siproxd.general.plugin_regex_replace='\10'


uci add_list siproxd.general.plugin_regex_desc='append_0'
uci add_list siproxd.general.plugin_regex_pattern='^(sips?:)([0-9]{10}@)'
uci add_list siproxd.general.plugin_regex_replace='\10\2'
uci commit
/etc/init.d/siproxd restart

The above commands are similar to adding the following to /etc/config/siproxd

Code:
list load_plugin 'plugin_regex.so'

# replace indian coutry code +91 with 0
list plugin_regex_desc   = 'country_code'
list plugin_regex_pattern = '^(sips?:)\+91'
list plugin_regex_replace = '\10'

# append 0 if dialed number is 10 digits
list plugin_regex_desc    = 'append_0'
list plugin_regex_pattern =  '^(sips?:)([0-9]{10}@)'
list plugin_regex_replace = '\10\2'
 
I find the built-in SIP client in Android to be better than GS Wave. It's simple and just works.

If you enable SIP calling for all calls, you can make calls from the native dialer just like choosing the SIM to use for the call.

View attachment 1265View attachment 1266
Hey @varkey , how did you do that?
I have a Netlink modem and I tried a lot to setup voip on my native dialer on Samsung, Xiaomi, and Realme. It just didn't work. Have to rely on zoiper which is highly unreliable.
 
Ok, so I figured it out. The issue is that, SIP doesn't work so well when behind NAT, that's the reason when there is NAT involved, it fails.

To fix that, I used something called siproxd which is installed on my OpenWRT router. This acts like a proxy and works around the NAT issues by modifying the SIP requests and stuff on the fly.

Now I am able to make and receive calls with any SIP client, in fact I am able to make calls from Bangalore through my BSNL VOIP connection in Kerala (I use Zerotier for the remote connectivity) 🕺
 
So I was thinking of the same thing the other day, and I think it's feasible. You would just need to add a custom route on your router to use the voice bridged interface to reach 192.168.149.0/24 or something.

On those Netlink ONT devices, you can bind a port for each WAN connection. So internet WAN connection should be bound to say LAN 1 and voice WAN connection should be bound to say LAN 2. You'd also a need a router with multiple WAN ports or configurable WAN ports like a OpenWRT device for example.

Then on your main router, the voice interface can be configured with static IP and also add a custom route for the SIP registration address. After this we ideally should be able to use any SIP application to make calls.

This of course would need the MAC address binding to be reset, but certainly is a great idea.
Could you please tell step by step procedure for using VoIP on android sip server is on 192.168.x.x gateway is 10.x.x.x. what change or route required to configure. Please reply I am new to this
 
Could you please tell step by step procedure for using VoIP on android sip server is on 192.168.x.x gateway is 10.x.x.x. what change or route required to configure. Please reply I am new to this

Try this. I have tried and working fine :

 
@varkey - Got this working.
I am using pfsense as firewall and siproxd was available for it as well.
Incoming and outgoing to landline numbers is working but outgoing to mobile numbers say "this facility is not available at this moment"
Hello

I am also using pfsense as firewall and siproxd.
can you share the steps and settings you used for pfsense as well as for siproxd.
i am using the ONT in bridge mode...Alphion c100 model.
 
Try this. I have tried and working fine :


Hello

Can you tell what to do if the ONT modem(GPON - Alphion C100) is in bridge mode and firewall/router is pfsense 2.6??
 
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