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I had the same exact problem with Bell in Quebec City. It went away when I changed the voip.ms server to montreal3.voip.ms instead of montreal4
 
Also, I've put Google's secondary DNS server as the primary DNS server of the Obihai as shown below. That helps as well !

AddressingType DHCP
IPAddress 192.168.0.85
SubnetMask 255.255.255.0
DefaultGateway 192.168.0.15
DNSServer1 8.8.4.4
DNSServer2 192.168.0.15
 
I had the same exact problem with Bell in Quebec City. It went away when I changed the voip.ms server to montreal3.voip.ms instead of montreal4

Odd.

I'm using the toronto server here (since Teksavvy's exit point used to be in Toronto back at the time).
 
Ok, I'm about to give up ... I have no clue what could be the problem. It's a one way audio problem and it's related to a NAT problem. My firewall is not enabled, the SIP passthrough is enabled, no port forward, no DMZ ...
 
Ok, I'm about to give up ... I have no clue what could be the problem. It's a one way audio problem and it's related to a NAT problem. My firewall is not enabled, the SIP passthrough is enabled, no port forward, no DMZ ...

Keep the router's firewall enabled, otherwise you might run into problems.

All I can recommend is doing a factory default reset on the ATA, and reconfigure it again following their Wiki. NAT handling is handled through some settings on the ATA.

https://wiki.voip.ms/article/Cisco_SPA112

You could also test with a different server. In the past, that ATA has had issues with older versions of Asterisk - it was another reason reason why I was using that specific toronto server, as at the time it was one of the two with a newer version of Asterisk.

Also, make sure you have no other SIP client running at home, just in case it could conflict with the Cisco.
 
UPDATE: my bad, I'm actually using chicago.voip.ms, not the Toronto server. It was the closest one to Toronto (where Teksavvy's termination is), and it was running a newer (at the time) Asterisk version than the Toronto server. I was having random registration issues with Toronto.

I'm waiting for them to deploy their new global voicemail system to try moving back to a Montreal-based server.

For additional info, my SPA112 runs firmware 1.4.1.
 
I don't think it's the server, because my ATA is now directly on the internet (with the PPPOE passthrough) and it's working. My SPA 112 is on 1.4.1 also. I'll try to enable the firewall.
 
Did you try setting up STUN on your ATA? I have a Grandstream ATA on voip.ms, I did that to solve one-way audio if I remember correctly.

I don't think it's the server, because my ATA is now directly on the internet (with the PPPOE passthrough) and it's working. My SPA 112 is on 1.4.1 also. I'll try to enable the firewall.
 
Go to the web interface page for your SPA112, I believe you have a SIP page in there? Look for something like "NAT Support Parameters" section, and look for fields to enter STUN Enable, STUN Test Enable, and STUN Server. You need to enable STUN and enter a STUN server DNS name, any working one will do. I used stun.stunprotocol.org:3478. You may be able to leave of the port at the end. Give it a go. Note I have a Grandstream HT701, not a Cisco SPA112, so not so familiar with yours but the Grandstream has basically the same settings as your Cisco.
 
Well...

I had voip.ms working before.... changing the setup ;(

Now the ONT is connected straight into the Asus as opposed when the Asus was connected into the HH2000. Somehow I got an external ip from the HH2000 and all was well but I suspected that I was loosing a bit of speed from the connection between the two.

I was right. I connected the ONT to the Asus and gained about 80 Mbits on download right there. I'm flirting with the 940 Mbits limit from Bell at 895 Mbits. BUT... I lost the voip.ms phone again.

Rebooted the Obihai to no avail.

What I'm suspecting, sorry RMerlin, is that the router is blocking packets going OUT since I can receive phone calls but can't get heard over the line... I'll have to dive into the protocols to understand better...

Will update the thread with findings, if any...
 
I also have Grandstream HT701 but do not have stun configure.

ht701.png


On voip.ms side:
voipms.png
 
Well...

I had voip.ms working before.... changing the setup ;(

Now the ONT is connected straight into the Asus as opposed when the Asus was connected into the HH2000. Somehow I got an external ip from the HH2000 and all was well but I suspected that I was loosing a bit of speed from the connection between the two.

I was right. I connected the ONT to the Asus and gained about 80 Mbits on download right there. I'm flirting with the 940 Mbits limit from Bell at 895 Mbits. BUT... I lost the voip.ms phone again.

Rebooted the Obihai to no avail.

What I'm suspecting, sorry RMerlin, is that the router is blocking packets going OUT since I can receive phone calls but can't get heard over the line... I'll have to dive into the protocols to understand better...

Will update the thread with findings, if any...
Ok so now you have the same setup as me, and you have the same problem, did you find out what is the problem ?

Thanks !
 
@Kronyx and @RMerlin ,

I was thinkering with iptables and inserted those rules in the FILTER section:

admin@Sarah:/tmp/home/root# iptables -A INPUT -p udp -m udp --dport 5060 -j ACCEPT
admin@Sarah:/tmp/home/root# iptables -A INPUT -p udp -m udp --dport 5080 -j ACCEPT
admin@Sarah:/tmp/home/root# iptables -A OUTPUT -p udp -m udp --dport 5060 -j ACCEPT
admin@Sarah:/tmp/home/root# iptables -A OUTPUT -p udp -m udp --dport 5080 -j ACCEPT

It didn't workout though but I think I might be onto something...
 
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