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 ?
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 ?