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biznet
biznet's picture
Joined: 2016/01/19
Points: 10

We are having issues After working fine, when Connect switches back to desk phone, the phone can dial out but goes directly to VM on incoming calls. From the below it appears to not be registering completely with the switch after switching back to desk phone.. Here is all the data I have about it.

phone ext 2003 is behind a nat router using wireless connection.

we just upgraded our switch to MT 8.3.1.2 . Tesing Thirdlane connect.

When fresh reboot and registration of phone unit, all is well reports status in PBX Snapshot :

2003-BNT/2003-BNT 216.252.192.171 D Yes Yes 5060 OK (24 ms

Connect registers first time and calls are forwared to cell (Android app) all is good.

### from logs after switch to Connect ###
[2018-03-20 12:48:50] VERBOSE[1791] chan_sip.c: -- Registered SIP '2003-BNT' at 216.252.192.171:5060
[2018-03-20 12:48:50] NOTICE[1791] chan_sip.c: Peer '2003-BNT' is now Reachable. (24ms / 2000ms)
[2018-03-20 12:59:05] VERBOSE[25700] chan_sip.c: -- Registered SIP '2003-BNT' at 127.0.0.1:48088

### from PBX Snapshot ####
2003-BNT/2003-BNT (Unspecified) D Yes Yes 0 UNKNOWN

#### called ext 2003 #### Call received

#### from PBX Snapshot after first call to app, received ####
2003-BNT/2003-BNT 127.0.0.1 D Yes Yes 48206 OK (319 ms)

### second call to 2003 #### worked same PBX snapshot

### switch back to desk from Connect android APP###

### from PBX Snapshot before any call testing ###
2003-BNT/2003-BNT (Unspecified) D Yes Yes 0 UNREACHABLE

#### from logs right after switch back from Connect
[2018-03-20 13:08:16] VERBOSE[26964] chan_sip.c: -- Unregistered SIP '2003-BNT'

#### attempted call to ext 2003 ####
Goes directly to VM, which I'd expect given registration is not shown

#### from Snapshot ###
[2018-03-20 13:08:16] VERBOSE[26964] chan_sip.c: -- Unregistered SIP '2003-BNT'

#### temp disconnect (wireless) the re-connect network still no registration
#### cold boot phone --all is well at Desk phone, incoming and outgoing calls.

Thanks, any ideas? Customers reported this problem before the update from 8.1.3 to 8.2.1.3 yesterday using Connect 1.4.1

mcampbell@1poin...
mcampbell@1pointcom.com's picture
Joined: 2015/05/20
Points: 0

Hi,

Make sure NAT Keepalive is set on the desk phone and ensure NAT is enabled on TL extension. This has fixed issues with OB calls working & IB calls failing for us behind NAT before; however I'm not sure whether this will help with TL Connect related issue.

biznet
biznet's picture
Joined: 2016/01/19
Points: 10

Thanks, Matt. I can't find any settings on the Yealink phone for Nat Keepalives, although it may be called something else. I'll put the phone on an outside address and see if that works, although I can't keep it there, it may tell me something.

Anyone else? Thirdlane?

biznet
biznet's picture
Joined: 2016/01/19
Points: 10

Update: From Matt's suggestion I put the phone on a non natted ip address and the same problems exactly persist.

would love to get Connect working consistently, it is a great feature, but buggy at least on our system HELP.

Doug

mcampbell@1poin...
mcampbell@1pointcom.com's picture
Joined: 2015/05/20
Points: 0

Hi Doug,

The keep alive setting on the Yealink can be found under Account --> Advanced. There are 2 settings on this phone you should pay attention to: #1 Keep Alive Type = Options and #2 Keep Alive Interval (Seconds) = 30

Based on your test with a non-NAT address it likely won't help, but it's worth a try...

biznet
biznet's picture
Joined: 2016/01/19
Points: 10

I changed Matt's Keep alive suggestion and when testing today it still does no work. Is anyone else at Thirdlane listening Alex, somebody?

eugene.voityuk
eugene.voityuk's picture
Joined: 2014/01/28
Points: 90

This has no any relation to NAT or Keepalives. The issue in nutshell is in the chan_sip... Asterisk chan_sip could have only one contact per AoR (one registered sip phone per account at a time), and only one set of settings per contact. This is why switching mechanism exists. When Connect switches to WebRTC mode, we restrict any register messages coming to Asterisk except thru WS per account. This blocks any registrations from TCP, UDP, TLS, and allows only Connect to register and don't fight for it right to be reachable, with the desk phone, which would not even be able to support all those ICE and DTLS-SRTP settings which are mandatory for Connect. Now, when you switching back to track your desk phone within Connect, your phone "does know nothing" about any switch happened, and we don't have any good way to tell this to your phone. So it should receive some poke, to re-register, or you should wait till next registration, based on your phone registration interval. This solution was applied as temporary, with the idea in mind, that people will not be switching back and forth all the time, and when someone will switch to the desktop phone, he will be responsible to poke the phone. We do realize that this is not an ideal solution and right after initial release we started a big project that aims to fix this and
the huge amount of related issues. Connect support for simultaneous login on multiple devices per single account alongside with one registration per sip account is planned for the next major release. This means that you will be able to have few Connect instances (home/work/mobile) and your desk phone working simultaneously. We will publish the blog post about next major release, that will explain all the differences, improvements, and benefits once it will be ready, before release.

biznet
biznet's picture
Joined: 2016/01/19
Points: 10

Ok, thanks for the explanation. this may help others. I will tell customers they need to "poke" the phone when they get switch back.

Doug

mattdarnell
mattdarnell's picture
Joined: 2007/10/25
Points: 20

Unless it is a very large system, I would set the phones to re-register every 60 seconds.

biznet
biznet's picture
Joined: 2016/01/19
Points: 10

Just found the place on the phone config to tell it to re-register every N seconds, thanks mattdarnal, this is the fix.