Calls not answering sporadically

[oliver.nash]

Hi all,

Firstly, apologies if this isn't supposed to be posted here.

I'm having an issue whereby calls frequently aren't answered. I.e., the call rings through to the correct extensions/group but when the handset is picked up the call just keeps ringing.


The phones in question are Yealink T56A's using the default provisioning template. I've setup syslog on one of the handsets and it appears that the handset is working correctly (i.e. is recognising the call has been picked up). But when I check the call log on the management console I see that 3CX is showing the call as 'Not Answered'.

I've checked this with the trunk provider and they've confirmed that everything from their end is configured correctly (it's VoiceHost so it's 3CX approved). I've also confirmed that the firewall is configured correctly and SIP ALG is disabled on the router in front of the handsets.

I'd be super grateful for any suggestions anyone has as to why this is happening. It doesn't appear to affect every call but certainly affects more than 50%.

Many thanks,

Ollie

 

[AWS2P]

Hi,
Please give us best context view answering this first
https://www.3cx.com/community/threads/information-to-provide-when-requesting-help.67558/

 

[oliver.nash]

Hi there,

Thanks for the link, here are the details:

• 3CX Version - Standard Annual 16.0.493
• Server OS - Debian 9.11
• Is the 3CX Server Hosted and where? - Hosted on Microsoft Azure
• IP Phone Make/Model/Firmware - Yealink T56A, Firmware 58.83.0.25
• Provisioning Method: Local / VPN / STUN / SBC - STUN
• Trunk Provider or Gateway Make/Model - VoiceHost
• Has the Firewall Checker passed: - Yes
• Are custom Phone Templates being used: - No

Kind regards,

Ollie

 

[Evolute IT]

Hi there,

Thanks for the link, here are the details:

• 3CX Version - Standard Annual 16.0.493
• Server OS - Debian 9.11
• Is the 3CX Server Hosted and where? - Hosted on Microsoft Azure
• IP Phone Make/Model/Firmware - Yealink T56A, Firmware 58.83.0.25
• Provisioning Method: Local / VPN / STUN / SBC - STUN
• Trunk Provider or Gateway Make/Model - VoiceHost
• Has the Firewall Checker passed: - Yes
• Are custom Phone Templates being used: - No

Kind regards,

Ollie

How many phones are in STUN?

Have you port forwarded the SIP and RTP ports to each phone using a reserved private IP?

 

[oliver.nash]

How many phones are in STUN?

Have you port forwarded the SIP and RTP ports to each phone using a reserved private IP?

All the phones are in STUN

Yes, IP is bound to the MAC of the handset and relevant ports are forwarded to the handset.

 

[Evolute IT]

All the phones are in STUN

Yes, IP is bound to the MAC of the handset and relevant ports are forwarded to the handset.

All phones, meaning? 2-3 or 25? STUN is not recommend for more than 2-3. The SBC might solve your issue there.

Have you assigned different ports for each phones?

 

[oliver.nash]

Apologies, it's 2 Yealink IP phones + 2 provisioned with STUN via the 3CX app.

Yes, different SIP and RTP are configured for each phone.

 

[Evolute IT]

Apologies, it's 2 Yealink IP phones + 2 provisioned with STUN via the 3CX app.

Yes, different SIP and RTP are configured for each phone.

Okay. Have you tried setting up an SBC to see if this fixes your issue?

Also, make sure SIP ALG is disabled in your firewall.

 

[JohnS_3CX]

Hi Oliver,

Is there any SIP-ALG running on the router at the phones site?

 

[oliver.nash]

Okay. Have you tried setting up an SBC to see if this fixes your issue?

Also, make sure SIP ALG is disabled in your firewall.

Thanks Frederick. I'm reluctant to set up an SBC at this stage purely because it's only one phone at each site (distributed team) so even if this works it's not really a viable long term solution. Will definitely give this a go if other troubleshooting yields no results!

Yep, SIP ALG is disabled on the firewall already.

 

[oliver.nash]

Hi Oliver,

Is there any SIP-ALG running on the router at the phones site?

Hi John,
No, SIP-ALG has always been disabled on the router at site.

 

[JohnS_3CX]

Hi Oliver,

As suggested above, try the SBC. I think it will take care of any delays or loss that may bring the invites "out of sync" lets say to the point where the phone is trying to pick up a call that no longer exists.

If the SBC keeps disconnecting, I think you might have to look at the network stability. Make sure TLS tunnel is on (default on SBC).

Otherwise I would look into time sync issues too (on both ends)

 

[oliver.nash]

Hi Oliver,

As suggested above, try the SBC. I think it will take care of any delays or loss that may bring the invites "out of sync" lets say to the point where the phone is trying to pick up a call that no longer exists.

If the SBC keeps disconnecting, I think you might have to look at the network stability. Make sure TLS tunnel is on (default on SBC).

Otherwise I would look into time sync issues too (on both ends)

Thanks John, much appreciated. I'll double check the time config on each end test before setting up the SBC just to see if that solves it.