I have a Axis A8105-E Intercom that partially works with Generic Sip. Help!

[mvasquez20]

Good afternoon,

I have an Axis A8105-E intercom door camera, and I would like to get it working with Zoom Phone. After entering in all the information for the manual provisioning, and enabling sips I was able to get it provisioned with Zoom phone. I can make calls to the common area extension assigned to the phone, with the device automatically picking up and two way audio established. However, when I press the button and attempt to have the device call a Zoom extension, or even an outside phone number, the call rings, and when the recipient picks up the call just ends.

Are there any settings I am missing? Any insight or advice would be greatly appreciated.

[IP-Man]

I would suggest that you log a problem with Zoom Support at the site below, so support team can help you with the concerns, make sure you have call IDs /call samples that you can provide them so they can further check.
https://support.zoom.us/hc/en-us/requests/new

[mvasquez20]

I took your advice and put a ticket in a moment ago. But this is a generic sip endpoint that is not on the supported hardware list. I am unsure how much support I will actually receive from Zoom phone support. 

[bayside]

Did you ever get this working? We have AXIS C1410 speakers. Haven't been able to get them to connect to Zoom. Support wouldn't help because it's not supported hardware. 

[stevenhuang]

I had the similar issue with my C1410 speaker, where it would disconnect the call right after connecting. It turns out to be a codec selection issue.

Try the following and see if it solves your issue:

• Under SIP settings -> Audio -> only select G.722, and unselect all others.
• Under SIP settings -> Ports -> change TLS port to 5091.
• Under SIP Accounts -> make sure to use TLS when registering to Zoom.

[mvasquez20]

I checked the device and I don't see an option for G722 audio. 

[bayside]

I tried your settings but  I still get a 408 timeout message.

Here's the settings I'm using in SIP Accounts

Transport=TLS

TLS Version=1.2

Media Encryption= SRTP Mandatory

Client Cert=default (Self signed)

Verify server certificate = checked

SIPS=checked

Proxy Transport= TLS

I've upgraded to latest firmware too.

Do my settings under SIP Accounts match yours?

[twiley]

Any luck with getting this to work I think I'm in the same spot you are.
Zoom gave me this:

Reason SIP ;cause=480 ;text="SIPS Required"

Axis tried to help and offered some suggestions such as setting DNS servers and NTP up.

[kwanlow]

See if you can unchecked the SIPS. ZP do not use SIPS.

[Eliot]

hi mvasquez20,

this appears to be a discontinued item.

AXIS A8105-E Flush Mount - Product support | Axis Communications

Accessories_Product Discontinuation Statement (axis.com)

maybe you might consider switching to zoom certified alternative such as 2n, algo or cyberdata?

Zoom Phone Certified Hardware

thanks,  eliot

[mvasquez20]

Unfortunately, Algo doesn't sell intercom cameras, and 2N cameras look not great based on the model we bought, and comments online. Checking Cyberdata's intercom, and the camera resolution is the same as 2N. Not seeing good door intercom options that are reasonably priced like the axis unit.

[Kayce]

Is it possible that your device was reasonable because it is a discontinued item? Have you checked Amazon or eBay for one of the supported items? Maybe you can get it cheaper through there.

[kwanlow]

Alternatively... please check out the Grandstrream GDS 3710 video intercom solution. It's on the supported device.

[Eliot]

hi twiley,

i suspect the error message may be caused by issues with security settings and certificates.

please see zoom support article for details and some certified devices.

Zoom Phone Interop-certified hardware

https://support.zoom.com/hc/en/article?id=zm_kb&sysparm_article=KB0074333

thanks,  eliot

[TheJourney]

Did this get resolved? We are trying an Axis I8016

[twiley]

No this didn't get resolved, I tried what I believe is the I8016 as well.  Yielded the same result

[TheJourney]

Ours is reg to Zoom, however calls disconnect when you answer in ZP. ZP can call our device, and two way audio works. But not from calls from the device, they hang up when ZP answers. 

[TheJourney]

We got our Axis I8016 working! Making calls and opening doors with ZP! 

[TroyWiley]

This is awesome care to share the details of how you set it up? Perhaps you would be open to a zoom call?

[TheJourney]

Sure ill try, hope this helps. I did hear from Zoom folks that Axis is working on being supported! 

I think our issue was old code on the device that did not have the Transport mode switch under Proxies. So after we updated the FW to the current version 12.0.98 and Turned that on TLS mode, unchecked SIPS. Also reboot after ANY change to the Axis device seems to be a good rule to follow. 

SIP ACCOUNT: 

Transport mode: TLS

TLS version: Auto

SRTP Mandatory

Client cert: axis device ID RSA-2048 

Verify Server cert: Checked

SIPS: UNCHECKED

Proxies

Transport: TLS

NAT: ICE is checked. 

Also we left opus codec on top, ZP likes opus... and don't send video to ZP. Video Direction is inactive. 

[TroyWiley]

That works for the I8016-LVE it does need the latest code.

[ZoomieKitty]

I've done all of those settings, phone is registered and I'm able to make a sip call to a SIP account I setup on Linphone, but when I put in the sip number for a zoom phone# to test, it fails right away. Not sure if I'm putting it in wrong. Is this the correct format.

12345678901 ATsymbol 102981 dot zoom dot us:5091

---

[ZoomieKitty]

I figured it out, I had to use the outbound proxy address not the domain.

[SilentDogood]

Can confirm... The following settings work for the Axis A8207-VE Mk ii   as well.

Main Settings

• Upgrade to the latest version  (version 12.0.98 min)
• SIP Settings
  • Ports SIP 5090, TLS 5091, RTP Start 20000
  • NAT Traversal :    ICE Enabled   (Turn/Stun disabled)
  • Audio and Video :   Opus Enabled (disable all others)
  • Video Direction:   INACTIVE
    
    

SIP Account Settings

• Transport Mode : TLS
• TLS Version : Auto
• Media Encryption: SRTP Mandatory
• Default Cert
• Verify Server Certificate:   Checkbox enabled
• NO CHECKBOX on SIPS
• Proxies,   Use TLS Transport