Is there VOIP service/SIP settings in AIRTEL xtreme fiber broadband

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21
Location
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ISP
BSNL
Hello all wanted to know is there any VOIP settings for AIRTEL xtreme fiber users if so how to access same using SIP settings ..thx
 
Can someone suggest me that if want to use an IP telephony with the above SIP Given settings how do I do the same? I still largely confused.
 
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@sumitroop On the telnet interface if you do a
Code:
user(show)# network interface all

it will list the ipv4 and subnet alloted for voice traffic like so

Code:
pon_v666_0_1   Link encap:Ethernet  HWaddr blahblah
               inet addr:10.146.xx.xx  Bcast:10.146.xx.xx  Mask:255.255.254.0
               UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
               RX packets:65014 errors:0 dropped:0 overruns:0 frame:0
               TX packets:65152 errors:0 dropped:0 overruns:0 carrier:0
               collisions:0 txqueuelen:0
               RX bytes:13469121 (12.8 MiB)  TX bytes:13761836 (13.1 MiB)

The gateway for the voice traffic is revealed by arp
Code:
user(show)# arp   

ARP ENTRIES IN CACHE:

  (10.146.158.1) at **redacted** [ether]  on pon_v666_0_1


This is the ip to use on your downstream modem untagged VLAN.

The sip server ip is show at Status>Voice Information

Code:
Soft Switch 10.232.139.146

Here is what i suggest you can try.

  1. Delete the inbuilt WAN profile, and create separate VLAN 100 and Voice VLAN 660 profiles (try with VLAN666 if 660 does not work)
  2. Bind LAN2 to the Voice profile with connection type IPOE and Bridge
  3. On the downstream modem or PC setup a static ip address with the details obtained from the above. Use the gateway as obtained from the arp.
  4. Setup static route 10.232.0.0 via voice gateway found above from arp (e.g. route add 10.232.0.0 mask 255.255.0.0 10.146.158.1)
  5. Check if you can ping to the softswitch at the ip address obtained above

@rajil.s when I try to create a profile with suggested option i.e. IPOE & WAN mode as Bridge WAN it gives error "The current service type is not supported in bridge WAN mode." I also saw the screenshot that you have enclosed earlier & the setting you mentioned there are working fine i.e. IPOE with Route WAN & IPOE.

Image enclosed by you in earlier post
Screenshot (14).webp
 
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I just change some settings told by sloj in his post-https://broadband.forum/t/203538/

1. Voice to voice internet (on Huawei modem it is disabled so I use to inspect element on the browser and enable via edit attribute)View attachment 3569
2. Enable NAT
3. Created Static route
4. Setup the Microsip like you guys did see in the attachment

View attachment 3566

before that I use Primary SIP address, So I was able to receive calls but can't call anyone, and so figure it out accidentally when I put the Standby proxy server address in the Proxy box of microsip.
View attachment 3570


but can't figured it out on android, I also tried Csipsimple on android but no luck

@nehalahmad15@gm could you please share what static route did you put on your system. As I have tried something however doesn't work me.
 


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Hello Nehalahmad15,

I followed the exact same steps that you have mentioned above and I have restarted my router.
On android I am using Calls - SIP VoIP Softphone app.
I am able to receive call as well as able to call others.

But I am facing internet connectivity issue when I have enabled VOIP_INTERNET.
Any idea how to get internet connectivity while in VOIP_INTERNET mode ?
 
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@sumitroop Please post your messages here rather than writing on my profile.

On my pfsense router (downstream of the bridged Huawei H8145V5 router ) I have the following static route,

1598858729963.png


The interface 'AIRTELVOIP' interface is simply another 'WAN' on the router which is set to the same ip address which the HUAWEI router had on its voip interface. I removed the static ip from the Huawei and instead set it up on the pfsense router.
 
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Hello Nehalahmad15,

I followed the exact same steps that you have mentioned above and I have restarted my router.
On android I am using Calls - SIP VoIP Softphone app.
I am able to receive call as well as able to call others.

But I am facing internet connectivity issue when I have enabled VOIP_INTERNET.
Any idea how to get internet connectivity while in VOIP_INTERNET mode ?
Hello bhakta98rta , I tried the same able to register with Calls - SIP VoIP Softphone app however while making calls it is saying forbidden
 
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@sumitroop Please post your messages here rather than writing on my profile.

On my pfsense router (downstream of the bridged Huawei H8145V5 router ) I have the following static route,

View attachment 3794

The interface 'AIRTELVOIP' interface is simply another 'WAN' on the router which is set to the same ip address which the HUAWEI router had on its voip interface. I removed the static ip from the Huawei and instead set it up on the pfsense router.
Hello Rajil.s , thanks a lot for your response. I was able to register with a soft phone app for android using the method suggested by bhakta98rta however calls are getting forbidden error. I am just now using your method on a TP-Link router. will keep you posted
 
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@rajil.s

I attached a TP-Link router (downstream) to my Huawei H8145V5 assigned the static ip of my voip to the interface tp link interface. Also have cloned the Mac interface of TP-Link to the Mac interface of Huawei H8145V5. Have also applied the static routes in tp-link (downstream) still not able to ping the sip proxy. Please suggest

static ip on tp-link & cloned Mac of interface Huawei H8145V5

Screenshot 2020-09-04 at 7.35.10 PM.png


added static route

Screenshot 2020-09-04 at 7.34.05 PM.png



Routing table

Screenshot 2020-09-04 at 7.34.32 PM.png



Ping response

Sumits-MacBook-Pro ~ % ping 10.232.130.170


PING 10.232.130.170 (10.232.130.170): 56 data bytes


Request timeout for icmp_seq 0


Request timeout for icmp_seq 1


92 bytes from 10.175.93.172: Destination Host Unreachable


Vr HL TOS Len ID Flg off TTL Pro cks Src Dst


4 5 00 5400 a546 0 0000 3f 01 87be 192.168.0.106 10.232.130.170





Request timeout for icmp_seq 2


92 bytes from 10.175.93.172: Destination Host Unreachable


Vr HL TOS Len ID Flg off TTL Pro cks Src Dst


4 5 00 5400 8247 0 0000 3f 01 aabd 192.168.0.106 10.232.130.170





Request timeout for icmp_seq 3


Request timeout for icmp_seq 4


Request timeout for icmp_seq 5


Request timeout for icmp_seq 6


Request timeout for icmp_seq 7


Request timeout for icmp_seq 8


Request timeout for icmp_seq 9


Request timeout for icmp_seq 10


Request timeout for icmp_seq 11


Request timeout for icmp_seq 12


Request timeout for icmp_seq 13


Request timeout for icmp_seq 14


Request timeout for icmp_seq 15


Request timeout for icmp_seq 16


Request timeout for icmp_seq 17


Request timeout for icmp_seq 18


92 bytes from 10.175.93.172: Destination Host Unreachable


Vr HL TOS Len ID Flg off TTL Pro cks Src Dst


4 5 00 5400 f5bb 0 0000 3f 01 3749 192.168.0.106 10.232.130.170
 
Last edited:
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@rajil.s , could you please explain your setup.. Are you using a Huawei H8145V5 and have connect a linux system directly via Lan. Installed a pfsense firewall on your linux system or have used a pfsense router with a Huawei H8145V5.

My setup is something like this Huawei H8145V5 > physical cable > TP-Link > WIFI > Mac

Huawei H8145V5 - voice profile setup as bridge & bonded to a LAN port

Screenshot 2020-09-05 at 1.50.06 AM.webp
 
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@rajil.s - the static routes are all set (and I'm able to ping the Airtel's SBC from my PC) - but there are issues with SIP working through a NAT firewall.

We need to use something like a siproxd to make it work. We cannot use STUN because this is not public network.
Hey there!
I've been trying to configure my Huawei router to allow SIP traffic to pass through but I can't seem to find the routing options in the router's configuration page. I believe you've successfully configured it and I was hoping you could maybe help me out a bit with your settings.
Thanks
 
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@tellanand Try YATE and MicroSIP. They have the feature to make it work with NAT. It's YATE deos it automatically but MicroSIP has an option called Allow IP Rewrite.
Hey! I've tried using YATE with Airtel's SIP servers but it keeps saying request time out. I believe you have added some routes in your browser and I was hoping you could help me out a bit.
Thanks
PS: I'm a new user so I can't send private messages
 
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