Sip trunk TDE600-Asterisk calls drop after 30 seconds

[Oknet]

I set up a trunk between TDE600 SIPGW-16 ad a Freepbx system (pjsip trunk)
All is working fine except calls drop after 30 seconds.

Sip connection is over a site to site VPN, where two sites have different subnet.

I'm able for now to get just the asterisk trace :

[pre] -- Channel PJSIP/TDE-0000000d left 'simple_bridge' basic-bridge <3a2ad60e-9338-4d4d-ace9-4be2660368ce>
-- Channel PJSIP/207-0000000c left 'simple_bridge' basic-bridge <3a2ad60e-9338-4d4d-ace9-4be2660368ce>
== Spawn extension (macro-dialout-trunk, s, 33) exited non-zero on 'PJSIP/207-0000000c' in macro 'dialout-trunk'
== Spawn extension (from-internal, 799397, 12) exited non-zero on 'PJSIP/207-0000000c'
-- Executing [h@from-internal:1] Macro("PJSIP/207-0000000c", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/207-0000000c", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/207-0000000c", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("PJSIP/207-0000000c", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'PJSIP/207-0000000c' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/207-0000000c'
-- PJSIP/207-0000000c Internal Gosub(crm-hangup,s,1) start
-- Executing [s@crm-hangup:1] NoOp("PJSIP/207-0000000c", "Sending Hangup to CRM") in new stack
-- Executing [s@crm-hangup:2] NoOp("PJSIP/207-0000000c", "HANGUP CAUSE: 16") in new stack
-- Executing [s@crm-hangup:3] ExecIf("PJSIP/207-0000000c", "0?Set(__CRM_VOICEMAIL=)") in new stack
-- Executing [s@crm-hangup:4] NoOp("PJSIP/207-0000000c", "MASTER CHANNEL: 1688538323.13 = 1688538323.13") in new stack
-- Executing [s@crm-hangup:5] GotoIf("PJSIP/207-0000000c", "0?return") in new stack
-- Executing [s@crm-hangup:6] Set("PJSIP/207-0000000c", "__CRM_HANGUP=1") in new stack
-- Executing [s@crm-hangup:7] AGI("PJSIP/207-0000000c", "agi://127.0.0.1/sangomacrm.agi") in new stack
-- <PJSIP/207-0000000c>AGI Script agi://127.0.0.1/sangomacrm.agi completed, returning 0
-- Executing [s@crm-hangup:8] Return("PJSIP/207-0000000c", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/207-0000000c'
-- PJSIP/207-0000000c Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=[/pre]

That seems just a normal clearing...
That asterisk has other pjsip trunks with other asterisk just working fine, so asterisk configuration seems ok
Should I configure NAT traversal on SIPGW16 ?
Other ideas ?

 

[obtsystems]

Calls between the Panasonic ant the asterisks system would be classified as trunk to trunk so check the trunk group doesn’t have a time restriction on it. It does by default of 10 minutes. But no harm testing

Other thing is if you have a Sonicwall for the vpn. You will have to push out the unsupervised udp stream timer on the vpn zone. It is the tcp one you need to do as it is looking for a tcp keep alive packet that is not sent and drops the stream. And I think that is 30 seconds by default

 

[Oknet]

Hi,
ACK was never received because of a double NAT involved.
Configuring NAT traversal on TDE does the trick !