One Extension times out when placing internal calls

[seantp]

I'm about to delete and recreate an extension with this problem, but before I do I'd like to understand what is going on if possible.

All extensions can call this extension. However, when this extension calls out to any internal extension it goes straight to VM. In the activity log its a sip timeout error

Latest version of 3cx Ent
Yealink T58 phones

I've reset this phones to factory twice. Looking through all the settings, can't figure out what would cause these besides a corrupt config in the phone server.

 

[seantp]

After deleting the extension I've realized it's not the server. Shortly after the other extensions I've been testing with behaving the same way, dialing an extension goes straight to voice mail without ringing the other extension.

The server is hosted in Azure with an IPSec VPN
The Gateway is a Sophos XG135
The switch is a Netgear GC752XP

I'm now wondering if there is something about the VPN configuration and the router that is causing problems.

 

[leejor]

I'm now wondering if there is something about the VPN configuration and the router that is causing problems.

I was going to ask what your set-up was, but you beat me to it. I think you are probably on the right track. See if the 3CX Activity Log is of any help. Are the sets seen as registered?

 

[seantp]

Yes, they are all registered. The activity log doesn't really help, just says the calls to the extensions time out. I can call the direct lines for each one just fine. It's internal ext to ext calls that have the problem.

 

[ACCPhone]

Have you tried PBX Delivers Audio?

 

[leejor]

Calls to voice-mail, external calls... are they OK? Compare a log of a DID call to the of an Extension call to the same line (perhaps try using a more detailed log), what's different? Have you tried placing a call using a 3CX client rather than the desk phones? How is your VPN network set-up? Do you share all (data and voice) on the same subnet?

 

[cobaltit]

Have you tried PBX Delivers Audio?

Doesn't appear to be an audio issue (RTP) but rather a SIP issue (signaling)

 

[seantp]

Finally had some time to come back to it. Had Sophos look at the router, and they just confirmed what I see from the TCP Dump, which is the phone server is having problems connecting back to the extension that is placing the call. What is weird is that is says ICMP destination unreachable, but I can ping the extensions from the phone server.


WireShark: Ext 100, IP 172.16.20.194, Call Source

((this goes on for a bit)
1806 34.939338 172.16.20.194 172.18.0.4 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x45CB2F60, Seq=12384, Time=40000)

1807 34.940254 172.16.20.194 172.18.0.4 SIP 682 Request: BYE sip:[email protected]:5060 |

1808 34.941389 172.16.20.194 172.18.0.4 RTCP 114 Receiver Report Source description

1830 35.068056 172.18.0.4 172.16.20.194 ICMP 170 Destination unreachable (Port unreachable)

1829 35.068040 172.16.20.194 172.18.0.4 RTCP 142 Sender Report Source description Goodbye

WireSharek: Ext 110, IP 172.16.20.192, Receiving Call

119 2.696674 172.16.20.192 172.18.0.4 UDP 46 5060 → 5060 Len=4
1173 29.899411 172.16.20.192 172.18.0.4 SIP 438 Status: 200 OK |
1477 32.694993 172.16.20.192 172.18.0.4 UDP 46 5060 → 5060 Len=4
1824 35.053114 172.16.20.192 172.18.0.4 SIP 438 Status: 200 OK |

From SIP Flows:

"172.16.20.194",""NA" <sip:[email protected]:5060>","<sip:[email protected]:5060>","SIP","00:00:25","11","COMPLETED","INVITE 407 200"

"172.16.20.194",""NA" <sip:[email protected]:5060>",""na" <sip:[email protected]:5060>","SIP","00:00:00","4","REJECTED","REGISTER 407"

 

[seantp]

After all that, I checked the receiving extension and it was still set to PBX delivers audio. After disabling that the call went through. I don't get it lol