Telnet won't help find the IP. Telnet does let you probe an IP (or name) to see if there is a possible SIP server reachable there.
To find the address you need to reach Telnor's internal phone DNS which must be able to resolve the name, as the other devices have it configured by name.
While connected in voip mode (wan 710 disabled) what does the routing table look like? Just type route.
I had WAN interface stopped and WAN6 is always on couldn't be stopped... do you need me to enable WAN and take another screenshot?
lan2 is guest interface - can be ignored I use this for dumb APs...
I didn't remember to take screenshot of interface when I switched to VoIP... here's an old one showing that VoIP is being used so that the picture above doesn't confuse you - it also tells me it's being used on the main status page of OpenWRT
ping 10.55.255.254
ping 10.234.31.49
nslookup asbcp.ims.telnor.rs
nslookup ims.telnor.rs
If the nslookups don't work try adding either DNS server (49 or 50) directly:
nslookup asbcp.ims.telnor.rs 10.234.31.49
etc.
nslookup -debug ims.telenor.rs
Server: 127.0.0.1
Address: 127.0.0.1:53
Query #0 completed in 5ms:
authoritative answer:
Query #1 completed in 8ms:
authoritative answer:
nslookup -debug asbcp.ims.telenor.rs
Server: 127.0.0.1
Address: 127.0.0.1:53
Query #0 completed in 4ms:
authoritative answer:
Query #1 completed in 7ms:
authoritative answer:
nslookup -debug ims.telenor.rs 10.234.31.49
Server: 10.234.31.49
Address: 10.234.31.49:53
Query #0 completed in 3ms:
authoritative answer:
Query #1 completed in 3ms:
authoritative answer:
nslookup -debug ims.telenor.rs 10.234.31.50
Server: 10.234.31.50
Address: 10.234.31.50:53
Query #0 completed in 3ms:
authoritative answer:
Query #1 completed in 3ms:
authoritative answer:
nslookup -debug asbcp.ims.telenor.rs 10.234.31.50
Server: 10.234.31.50
Address: 10.234.31.50:53
Query #0 completed in 3ms:
authoritative answer:
Name: asbcp.ims.telenor.rs
Address: 10.24.20.218
Name: asbcp.ims.telenor.rs
Address: 10.24.28.218
Query #1 completed in 3ms:
authoritative answer:
nslookup -debug asbcp.ims.telenor.rs 10.234.31.49
Server: 10.234.31.49
Address: 10.234.31.49:53
Query #0 completed in 2ms:
authoritative answer:
Name: asbcp.ims.telenor.rs
Address: 10.24.20.218
Name: asbcp.ims.telenor.rs
Address: 10.24.28.218
Query #1 completed in 3ms:
authoritative answer:
ping 10.55.255.254
PING 10.55.255.254 (10.55.255.254): 56 data bytes
64 bytes from 10.55.255.254: seq=0 ttl=64 time=2.323 ms
64 bytes from 10.55.255.254: seq=1 ttl=64 time=2.304 ms
64 bytes from 10.55.255.254: seq=2 ttl=64 time=2.107 ms
64 bytes from 10.55.255.254: seq=3 ttl=64 time=2.230 ms
^C
--- 10.55.255.254 ping statistics ---
4 packets transmitted, 4 packets received, 0% packet loss
round-trip min/avg/max = 2.107/2.241/2.323 ms
ping 10.234.31.50 and ping 10.234.31.49 don't work
I did this from openwrt shell , also had to tick this box in order to get authorative answers from nslookup, it kept giving me
Non-authoritative answer:
............................................................................................................................
Trying to add abscp.ims.telenor.rs to routing doesn't work , not sure what's wrong with the IP I got ?
The things that are now known on the voip network are asbcp at 10.24.20.218 and the dns servers at 10.234.31.49 and 50.
I don't know exactly how proxying works but maybe if you point your SIP client to asbcp as the proxy, the proxy server will look up the actual ims.telenor.rs
Add asbcp as a static route (/32 means this route is for a single IP address)
target 10.24.20.218/32
interface VoIP
gateway 10.55.255.254
Then you can uncheck Default Route and DNS on wan711 and return to normal wan default through wan710 so Internet connection works. Also it should now be possible to lookup (from the local DNS host entry) and ping asbcp.ims.telenor.rs while connected to the LAN. From there try configuring a SIP client.
Thank you !
I'm able to place calls now , not able to hear anything on either end tho 
I'm gonna have to figure out why. SIP client registers alright - I can receive and make calls but that's as far as it goes...
I've heard that ISP uses STUN server so I might need to set that up first in order to have audio?
STUNEnable = true
STUNServerAddress = 85.162.11.196
STUNServerPort = 3478
STUNUsername = Serial number
STUNPassword = should be inside router config file
( But I don't see any of these inside my router *I checked the config file... If my setup doesn't use STUN server , could it be firewall? )
My SIP client doesn't seem to be facing packet loss (0/0 packets ) ? Why no sound is going through idk...
Tried all audio protocols that are available and only these two G729/8000/1, PCMA/8000/1 "work"
This is route from ISP router btw, in case that might be helpful... If anyone has any ideas what else I could look for on it let me know...
The audio packets must be going to another server. You'll need to route to that server through the voip wan.
Run a tcpdump on UDP packets coming from the phone. After dialing there will be a stream of UDP carrying the voice.
Maybe try another codec?
It's not available in this SIP client, there's another one on fdroid but it has even less protocols 
Ok 
How would you do this on PC for example?
Open terminal , type tcpdump eno1 and look at the output on screen?
( I can do the same on phone I think, by giving root access to termux and doing tcpdump on wlan interface? )
Setting up everything to be able to capture packets from phone on router ought to be a pain ( 
)
On router CLI run tcpdump -i br-lan udp and host phone's LAN IP
tcpdump is not available on the router...
I already did it on phone tho : https://pastebin.com/GGqB78yj (M2012K11AG.lan is the phone)
I answered the phone at 02:28:46 , that's where audio should start
You can search for "ringing" with find text to see when I dialed the number
I'll try to figure out how to get tcpdump on my router hmm
Edit 1:
Installed tcpdump but this is happening xd
tcpdump -i br-lan udp host 192.168.1.25
tcpdump: 'udp' modifier applied to host
Edit 2:
My bad wrong command...
tcpdump -i br-lan and udp host 192.168.1.25 works
The voice packets were sent to 10.24.24.166(*), so you need to include it in the route to the voip wan. Presently they are sent to the wan710 and lost.
Since there aren't any real Internet sites in the 10.0.0.0 block, you could route almost all of that block to voip wan 711 and not lose anything, except that your wan IP is also in the 10 range. Expand the existing route that covers the SIP server to all of 10.24.0.0/16 should work.
(*) This IP is likely to be different on each call, within some range of the servers they have.
As far as I see from like last 4 calls - the only thing that changed is the last number:
10.24.24.166
10.24.24.164
10.24.24.162
Would this work?
For a /16 netmask, the last 2 numbers need to be 0, that is 10.24.0.0/16.
This overlaps the other existing route to the SIP server (10.24.X.X are all covered by the new route) so there is no need to have both.
Ahhh I see , btw IT WORKS !!!
TY soooooo much 
You're life saver , if it's not a bother could you tell me what to use instead of /16 so I can tell it to only change last digit?
I tried to be smart and did it like so You think it's good?
This is the chart I borrowed
That should work. Though it's not a problem making the routes wider than they need to be, since there is nothing in 10.24.0.0 on the regular wan 710.
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