Extension to Extension Hangup

I have a site with a FPBX Distro system hosted on Vultr.

Site works fine both incoming and outbound calls.

At this site I have a need for an ATA (OBI 200) to support a legacy PSTN device. The OBI works fine for outbound calls and registers with the PBX – no issues.

The problem I am having is that i also need the ability to have calls placed from the OBI extension (x111) to another internal extension (X101) just by dialing the extension number. When 111 the OBI dials 101 (a Yealink desk phone) the call connects and 101 rings – but then when 101 picks up I get an immediate disconnect.

I checked SNGREP and the logs and nothing looks off –

Guess it could be a signaling issue – but not sure why regular outbound calls to the PSTN would work but not internal calls?

Any ideas how to tackle this?

thanks

Assuming pjsip extensions, try setting (for both)
Rewrite Contact: Yes
Force rport: Yes
Direct Media: No
and confirm that anything NAT related is turned off on both devices.

If you still have trouble, at the Asterisk command prompt type
pjsip set logger on
make a failing test call, paste the relevant section of the Asterisk log at pastebin.freepbx.org and post the link here.

Might this be the issue

[2020-08-07 15:30:57] VERBOSE[10931] res_pjsip_logger.c: <— Transmitting SIP request (496 bytes) to TLS:69.121.94.192:20766 —>
BYE sips:[email protected]:20766;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 149.28.46.73:5061;rport;branch=z9hG4bKPj2e07c10b-1187-4262-97fe-fa976db9c43c;alias
From: sips:[email protected];tag=d67b7300-6ce8-4b74-a564-a02eed6fb937
To: sips:[email protected];tag=SP1abf313417fbaaa72
Call-ID: [email protected]
CSeq: 30783 BYE
Warning: 381 SIP “SIPS Required”
Max-Forwards: 70
User-Agent: FPBX-15.0.16.72(16.11.1)
Content-Length: 0

[2020-08-07 15:30:57] VERBOSE[5031] res_pjsip_logger.c: <— Received SIP response (499 bytes) from TLS:69.121.94.192:20766 —>
SIP/2.0 200 OK
Call-ID: [email protected]
CSeq: 30783 BYE
Content-Length: 0
From: sips:[email protected];tag=d67b7300-6ce8-4b74-a564-a02eed6fb937
To: sips:[email protected];tag=SP1abf313417fbaaa72
Via: SIP/2.0/TLS 149.28.46.73:5061;branch=z9hG4bKPj2e07c10b-1187-4262-97fe-fa976db9c43c;alias;received=149.28.46.73;rport=5061
Server: OBIHAI/OBi200-3.2.2.5921
X-RTP-Stat: PS=0,OS=0,PR=0,OR=0,PL=0,JI=0,DU=0,EN=G711U,DE=N/A,MOS=0.00

[2020-08-07 15:30:57] VERBOSE[1032][C-00000030] bridge_channel.c: Channel PJSIP/110-00000068 left ‘simple_bridge’ basic-bridge
[2020-08-07 15:30:57] VERBOSE[1032][C-00000030] app_macro.c: Spawn extension (macro-dial-one, s, 56) exited non-zero on ‘PJSIP/110-00000068’ in macro ‘dial-one’
[2020-08-07 15:30:57] VERBOSE[1032][C-00000030] app_macro.c: Spawn extension (macro-exten-vm, s, 26) exited non-zero on ‘PJSIP/110-00000068’ in macro ‘exten-vm’
[2020-08-07 15:30:57] VERBOSE[1032][C-00000030] pbx.c: Spawn extension (ext-local, 101, 3) exited non-zero on ‘PJSIP/110-00000068’
[2020-08-07 15:30:57] VERBOSE[1032][C-00000030] pbx.c: Executing [[email protected]:1] Macro(“PJSIP/110-00000068”, “hangupcall,”) in new stack
[2020-08-07 15:30:57] VERBOSE[1032][C-00000030] pbx.c: Executing [[email protected]:1] GotoIf(“PJSIP/110-00000068”, “1?theend”) in new stack
[2020-08-07 15:30:57] VERBOSE[1032][C-00000030] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2020-08-07 15:30:57] VERBOSE[1032][C-00000030] pbx.c: Executing [[email protected]:3] ExecIf(“PJSIP/110-00000068”, “0?Set(CDR(recordingfile)=)”) in new stack
[2020-08-07 15:30:57] VERBOSE[1032][C-00000030] pbx.c: Executing [[email protected]:4] NoOp(“PJSIP/110-00000068”, "PJSIP/101-0000006a montior file= ") in new stack
[2020-08-07 15:30:57] VERBOSE[1032][C-00000030] pbx.c: Executing [[email protected]:5] GotoIf(“PJSIP/110-00000068”, “1?skipagi”) in new stack
[2020-08-07 15:30:57] VERBOSE[1032][C-00000030] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2020-08-07 15:30:57] VERBOSE[1032][C-00000030] pbx.c: Executing [[email protected]:7] Hangup(“PJSIP/110-00000068”, “”) in new stack
[2020-08-07 15:30:57] VERBOSE[1032][C-00000030] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/110-00000068’ in macro ‘hangupcall’
[2020-08-07 15:30:57] VERBOSE[1032][C-00000030] pbx.c: Spawn extension (ext-local, h, 1) exited non-zero on ‘PJSIP/110-00000068’
[2020-08-07 15:30:57] VERBOSE[1039][C-00000030] bridge_channel.c: Channel PJSIP/101-00000069 left ‘simple_bridge’ basic-bridge
[2020-08-07 15:30:57] VERBOSE[10931] res_pjsip_logger.c: <— Transmitting SIP request (462 bytes) to TLS:69.121.94.192:47953 —>
BYE sip:[email protected]:47953;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 149.28.46.73:5061;rport;branch=z9hG4bKPj08bd0840-f260-42cf-8b97-b64ca2ad8f7f;alias
From: “Kingsbridge Front Gate” sip:[email protected];tag=024070f3-8137-424b-b747-09ce23c81a29
To: sip:[email protected];tag=1576626174
Call-ID: b3ef80a6-f13b-4e7d-915e-84a1e43b5c3f
CSeq: 12283 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: FPBX-15.0.16.72(16.11.1)
Content-Length: 0

[2020-08-07 15:30:57] VERBOSE[5031] res_pjsip_logger.c: <— Received SIP response (387 bytes) from TLS:69.121.94.192:47953 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 149.28.46.73:5061;rport=5061;branch=z9hG4bKPj08bd0840-f260-42cf-8b97-b64ca2ad8f7f;alias
From: “Kingsbridge Front Gate” sip:[email protected];tag=024070f3-8137-424b-b747-09ce23c81a29
To: sip:[email protected];tag=1576626174
Call-ID: b3ef80a6-f13b-4e7d-915e-84a1e43b5c3f
CSeq: 12283 BYE
User-Agent: Yealink SIP-T46S 66.85.0.5
Content-Length: 0

The SIPS Required says why it sent the BYE, but we need to see what came before. Please paste the log for the entire call (starting from when the original invite was received) at pastebin.freepbx.org and post the link here.

Also, please explain the OBi configuration. Your original post said it was on x111, but the present log shows x110. Is it registering on multiple SPx?

Sorry – I have 3 separate OBI 200 devices each with a different extension (110,111 and 112) at the site and each one connects to a single legacy PSTN device – all 3 OBIs are connected to 1 FPBX system

All 3 OBI devices exhibit the same extension to extension behavior but work fine when dialing out to external numbers over the PBX trunk

Here is a link to the pastebin: https://pastebin.freepbx.org/view/ad3bef44

thank you for helping with this

Is there a reason that you are trying to use TLS (port 5061) ?

Yes

The OBI connects to the server and makes outbound calls fine all using TLS

Can you explain further your thought?

Sorry, I still don’t understand why the BYE, but noticed that the OBi is doing NAT mapping which may be confusing the SIP stack. Try setting X_DiscoverPublicAddress, STUNEnable and ICEEnable all off.

If this doesn’t help, paste a new log and also report:
Does *43 (echo test) work correctly from the OBi (ensure that it’s not intercepting the * code)?
Does *43 work correctly from x101?
Do calls from one OBi to another work correctly? Calls from one Yealink to another? Calls from a Yealink to an OBi?

Because your call is being BYE’d because SIPS was required but not provided ?

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