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

I had previously mentioned that VoIP worked only when I made BSNLs default VoIP gateway as the default route on my Pi. This barred me from having internet access on Pi. I solved the issue by adding a static route to BSNLs default VoIP gateway like so:

ip route add 10.0.0.0/8 via 10.81.104.1

Using tcpdump I observed that BSNL uses multiple VoIP servers and all servers are on 10.0.0.0/8 network.

I'm not using NAT as siproxd does proxying that is similar to NAT
 
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) 🕺
I don’t actually have an OPENWRT router. Is there anything else I can do?
 
Hello All,

I have got a new BSNL FTTH connection and that has been installed and activated, Voice and Data works perfectly fine using Syrotech ONU (Wi-Fi, POT, 2 LAN). Now what I am trying to configure or rather trying to use my Grandstream GXP1625 IP Phone directly instead of the traditional analog phone. I have tried few configuration as mentioned earlier in the thread but no go. Can someone please summarize the entire settings on how to get the BSNL Voice work on a grandstream phone directly.
TIA

Regards,
Abhijit
Can you share a picture of the ONU as well as the admin dashboard?
 
I have BSNL Bharat Fiber 777 Plan from an LCO in Trivandrum, Kerala.
I was previously using a basic EPON ONU modem of a local ISP based in Trivandrum ( Shine Broadband).
While changing to BSNl bharat fiber i didnt change my modem but it doesn't have any ports for telephony.

So my doubt is , is it possible to configure calling using a voip software or something?
 
I've got this working in my IP EPABX.

Steps followed as below in Digisol router.

1. Deleted voice profile set by bsnl.
2. Created new wan connection in bridge mode with vlan as 1830 and bind it to lan1.
3. Lan1 is connected to lan port of my hardware EPABX. It has 2 lan ports so it was easy for me to change ip of my 2nd lan port even while maintaining connectivity with my home network.
4. I configured ip address of my ippbx as mentioned in sip profile of my router.
5. Then added trunk with bsnl ftth address & login details & voila, sip trunk was up.

inbound & outbound Calls working fine. Voice is clear. Only issue is there is cross connection sometimes.

Best part is i can use the trunk even for multiple simultaneous calls. Tried with 3 calls.


Thanks to this forum where I could learn about how to go through.
 


I've got this working in my IP EPABX.

Steps followed as below in Digisol router.

1. Deleted voice profile set by bsnl.
2. Created new wan connection in bridge mode with vlan as 1830 and bind it to lan1.
3. Lan1 is connected to lan port of my hardware EPABX. It has 2 lan ports so it was easy for me to change ip of my 2nd lan port even while maintaining connectivity with my home network.
4. I configured ip address of my ippbx as mentioned in sip profile of my router.
5. Then added trunk with bsnl ftth address & login details & voila, sip trunk was up.

inbound & outbound Calls working fine. Voice is clear. Only issue is there is cross connection sometimes.

Best part is i can use the trunk even for multiple simultaneous calls. Tried with 3 calls.


Thanks to this forum where I could learn about how to go through.
What you're doing here might be illegal or violation of terms of service. Be careful brother.
 
I have Netlink HG323RGW GPON Modem provided by BSNL LCO. Please help me to configure SIP on OpenWRT(DSL-2750U_1.0) Siproxyd/Asterisk.

If possible kindly provide screenshots of configuration from both from Netlink HG323RGW and OpenWRT(DSL-2750U_1.0)
 
hello , I am using excitel connection in bangalore and was able to configure bsnl wings in my router . i am getting incoming calls but not able to dial. . i think i have to make some changes in my voip basic / advanced settings . Please help
 
Can anyone really help me in configuring sip. I have installed the gransteam wave app. What should i enter for sip server ,sip user id ,sip authentication id ,password, fields etc. I am newbie here . If anyone can help me out please..
 
Just adding it here for my own future reference 😅

In GS Wave, you could use the following settings

Code:
SIP Server: 192.168.149.78 or kl.ftthvoip.bsnl.in (Please note only one of these would work, depending on how your VOIP account is provisioned)
SIP User Id and authentication Id: 91<std code><voip number> (for example 9148429xxxxxx)
Password: 0000 (yes it's stupid)

Screenshot_20200425-155056.png

However, please note that, to get calls to work properly, you need to install siproxd on your router. From a siproxd perspective, the config is really simple, all you need to give is the inbound and outbound interface names. In OpenWRT this is how it would be, where lan is as the name suggests the LAN network, you wouldn't have to change this. And wan_voip is the VOIP interface, this you would need to change as necessary.

Code:
    option interface_inbound lan
    option interface_outbound wan_voip

You'd also need to add static routes to that the VOIP traffic goes via the VOIP interface and not through the main internet gateway. In addition to the route to the SIP server, you'd also need to add the route for the destination 10.191.0.0/16 to go via the VOIP interface.

Screenshot 2020-05-12 at 6.16.42 PM.webp

Now in GS Wave, you need to set the IP address of your router where siproxd runs as an outbound proxy.

Screenshot_20200425-155104.jpg

With this, you would be able to make and receive calls as required.

You could also use the built-in SIP client in Android, screenshots from my phone below.

Screenshot_20200425-155119.pngScreenshot_20200425-155125.png

@philip
 
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I will try this with airtel, although when i called airtel for the sip password, they had no clue what i was taking about. They were like do you need the password for static ip address . I said no, sip stands for session initiation protocol for voip. ineed the username and password for that just like you have for pppoe. And they had no clue. although all the settings for SIP are there in my ONT just the password is hidden , if i can extract the password from the ONT then i think this can be done on airtel too.
 

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