Not able to do out&in calls

I have system & networking engineering background but im new to freepbx and voip technology.
im trying to setup freepbx with my yeastar TG gateway on SIP accounts trunks, but somehow after alot of research im not able to receive or make calls, i get error 503 sip unavailable after 10 seconds of trying to make a call

i did setup the trunk with the following config:
Trunk Name: FROM_8000_FOR_EXT1000
Outbound CallerID: 96654xxxxxxx
CID Options: Allow any
Maximum Channels: 1 (GSM numbers)

Dial Number Manipulation Rules:

match: 966N.

PJSIP Settings

Username & Auth username: 8000 (SIP account) on GSM gateway
Authentication: Outbound
Registration: Send
SIP server: TG800.mdns.example
Port: 5060 UDP
Context: from-pstn-toheader
DTMF Mode: Auto
From Domain : Tg8001.mdns.example
From User: 8000
Trust RPID/PAI: yes
Send RPID/PAI: PAI
Force rport: Yes

Outbound Routes

Route CID: <96654xxxxxxx>
Trunk Sequence for Matched Routes: FROM_8000_FOR_EXT1000

Dial Patterns that will use this Route

Match pattern: 966N.
Callerid: 1000 (im trying to setup 1 ext = 1 number for in and out calls) so only ext=1000 can use this trunk

Inbound Routes

Route: FROM_TG_TO_EXT100

DID Number: 100
Alert Info: none
Set Destination: Ext 100

SIP Settings

External Address: Tg800.mdns.example
Allow Anonymous Inbound SIP Calls: NO
Allow SIP Guests: NO
RTP Port Ranges: 10000-20000
RTP Checksums: YES
Strict RTP: YES
Codecs: Ulaw & Alaw

Misc PJSip Settings

Caller ID into Contact Header: NO

0.0.0.0 (udp)

Port to Listen On: 5060
Domain the transport comes from: tg801.mdns.example

Extension: 1000

DTMF Signaling: RFC 4733
Context: from-internal
Trust RPID: YES
Send Connected Line: YES
Media Use Received Transport: YES
Force RTP: YES

GSM gateway:
DMZ = ON
MDNS = ON (tg800.mdns.example)
NAT= OFF
DTMF: 3566
Firewall: allow 5060 udp and 10000-20000 udp to Freepbx

im trying to setup freepbx with 5 trunks 1 special trunk for each ext out and in calls

Log files after failing test call:

[2022-03-02 14:51:39] VERBOSE[2242] res_pjsip_logger.c: <--- Received SIP response (879 bytes) from UDP:51.211.79.78:5060 --->  

77848   SIP/2.0 183 Session Progress    

77849   Via: SIP/2.0/UDP 51.211.x.x:5060;branch=z9hG4bKPj72ca65bb-1c13-492c-964f-ffe4c501e72b;received=70.34.xx.xx;rport=5060  

77850   From: <sip:[email protected]>;tag=257d99e7-284b-4504-92a0-89a092f28eb5    

77851   To: <sip:[email protected]>;tag=as3334c587    

77852   Call-ID: 482ca8dc-2b65-4a95-8f30-9d2843d60815  

77853   CSeq: 23182 INVITE  

77854   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO    

77855   Supported: replaces, timer  

77856   Require: timer  

77857   Session-Expires: 1800;refresher=uas

77858   Contact: <sip:[email protected]>  

77859   Content-Type: application/sdp  

77860   Content-Length: 264

77861      

77862   v=0

77863   o=root 1778613766 1778613766 IN IP4 192.168.100.69  

77864   s=Asterisk PBX 1.6.2.6  

77865   c=IN IP4 192.168.100.69

77866   t=0 0  

77867   m=audio 19824 RTP/AVP 0 8 101  

77868   a=rtpmap:0 PCMU/8000    

77869   a=rtpmap:8 PCMA/8000    

77870   a=rtpmap:101 telephone-event/8000  

77871   a=fmtp:101 0-16

77872   a=ptime:20  

77873   a=sendrecv  

77874      

77875   [2022-03-02 14:51:39] VERBOSE[3382][C-00000021] app_dial.c: PJSIP/FROM_8000_FOR_EXT1000-00000040 is making progress passing it to PJSIP/1000-0000003f  

77876   [2022-03-02 14:51:39] VERBOSE[2455] res_pjsip_logger.c: <--- Transmitting SIP response (874 bytes) to UDP:213.57.179.91:53516 --->  

77877   SIP/2.0 183 Session Progress    

77878   Via: SIP/2.0/UDP 192.168.1.21:53516;rport=53516;received=213.57.179.91;branch=z9hG4bK-524287-1---cff44688f60fbc60  

77879   Call-ID: Hu46S0_cjNe8yJizUhBxYw..  

77880   From: <sip:[email protected]>;tag=e2092367    

77881   To: <sip:[email protected]>;tag=663217ec-16a9-41eb-9f6f-7756f5f17ef7  

77882   CSeq: 2 INVITE  

77883   Server: FPBX-16.0.15(16.20.0)  

77884   Contact: <sip:51.211.x.x:5060>  

77885   Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER  

77886   P-Asserted-Identity: "CID:1000" <sip:[email protected]>    

77887   Content-Type: application/sdp  

77888   Content-Length: 250

77889      

77890   v=0

77891   o=- 69146120 3 IN IP4 51.211.xx.xx  

77892   s=Asterisk  

77893   c=IN IP4 51.211.xx.xx  

77894   t=0 0  

77895   m=audio 14734 RTP/AVP 0 8 101  

77896   a=rtpmap:0 PCMU/8000    

77897   a=rtpmap:8 PCMA/8000    

77898   a=rtpmap:101 telephone-event/8000  

77899   a=fmtp:101 0-16

77900   a=ptime:20  

77901   a=maxptime:150  

77902   a=sendrecv  

77903      

77904   [2022-03-02 14:51:47] VERBOSE[2242] res_pjsip_logger.c: <--- Received SIP response (492 bytes) from UDP:51.211.xx.xx:5060 --->  

77905   SIP/2.0 480 Temporarily unavailable

77906   Via: SIP/2.0/UDP 51.211.xx.xx:5060;branch=z9hG4bKPj72ca65bb-1c13-492c-964f-ffe4c501e72b;received=70.34.xx.xx;rport=5060

77907   From: <sip:[email protected]>;tag=257d99e7-284b-4504-92a0-89a092f28eb5    

77908   To: <sip:[email protected]>;tag=as3334c587    

77909   Call-ID: 482ca8dc-2b65-4a95-8f30-9d2843d60815  

77910   CSeq: 23182 INVITE  

77911   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO    

77912   Supported: replaces, timer  

77913   Content-Length: 0  

77914      

77915      

77916   [2022-03-02 14:51:47] VERBOSE[3256] res_pjsip_logger.c: <--- Transmitting SIP request (433 bytes) to UDP:51.211.xx.xx:5060 --->

77917   ACK sip:[email protected]:5060 SIP/2.0    

77918   Via: SIP/2.0/UDP 51.211.xx.xx:5060;rport;branch=z9hG4bKPj72ca65bb-1c13-492c-964f-ffe4c501e72b  

77919   From: <sip:[email protected]>;tag=257d99e7-284b-4504-92a0-89a092f28eb5    

77920   To: <sip:[email protected]>;tag=as3334c587    

77921   Call-ID: 482ca8dc-2b65-4a95-8f30-9d2843d60815  

77922   CSeq: 23182 ACK

77923   Max-Forwards: 70    

77924   User-Agent: FPBX-16.0.15(16.20.0)  

77925   Content-Length: 0  

77926      

77927      

77928   [2022-03-02 14:51:47] VERBOSE[3382][C-00000021] app_dial.c: No one is available to answer at this time (1:0/0/0)    

77929   [2022-03-02 14:51:47] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:37] NoOp("PJSIP/1000-0000003f", "Dial failed for some reason with DIALSTATUS = NOANSWER and HANGUPCAUSE = 19") in new stack

77930   [2022-03-02 14:51:47] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:38] GotoIf("PJSIP/1000-0000003f", "0?continue,1:s-NOANSWER,1") in new stack

77931   [2022-03-02 14:51:47] VERBOSE[3382][C-00000021] pbx_builtins.c: Goto (macro-dialout-trunk,s-NOANSWER,1)

77932   [2022-03-02 14:51:47] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:1] NoOp("PJSIP/1000-0000003f", "Dial failed due to trunk reporting NOANSWER - giving up") in new stack

77933   [2022-03-02 14:51:47] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:2] Progress("PJSIP/1000-0000003f", "") in new stack    

77934   [2022-03-02 14:51:47] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:3] Playback("PJSIP/1000-0000003f", "number-not-answering,noanswer") in new stack  

77935   [2022-03-02 14:51:47] VERBOSE[3382][C-00000021] file.c: <PJSIP/1000-0000003f> Playing 'number-not-answering.ulaw' (language 'en')  

77936   [2022-03-02 14:51:47] VERBOSE[2455] res_pjsip_logger.c: <--- Transmitting SIP response (874 bytes) to UDP:213.57.xx.xxx:53516 --->  

77937   SIP/2.0 183 Session Progress    

77938   Via: SIP/2.0/UDP 192.168.1.21:53516;rport=53516;received=213.57.xx.xx;branch=z9hG4bK-524287-1---cff44688f60fbc60    

77939   Call-ID: Hu46S0_cjNe8yJizUhBxYw..  

77940   From: <sip:[email protected]>;tag=e2092367

77941   To: <sip:[email protected]>;tag=663217ec-16a9-41eb-9f6f-7756f5f17ef7  

77942   CSeq: 2 INVITE  

77943   Server: FPBX-16.0.15(16.20.0)  

77944   Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER  

77945   Contact: <sip:51.211.xx.xx:5060>    

77946   P-Asserted-Identity: "CID:1000" <sip:[email protected]>    

77947   Content-Type: application/sdp  

77948   Content-Length: 250

77949      

77950   v=0

77951   o=- 69146120 3 IN IP4 51.211.xx.xx  

77952   s=Asterisk  

77953   c=IN IP4 51.211.xx.xx  

77954   t=0 0  

77955   m=audio 14734 RTP/AVP 0 8 101  

77956   a=rtpmap:0 PCMU/8000    

77957   a=rtpmap:8 PCMA/8000    

77958   a=rtpmap:101 telephone-event/8000  

77959   a=fmtp:101 0-16

77960   a=ptime:20  

77961   a=maxptime:150  

77962   a=sendrecv  

77963      

77964   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:4] Congestion("PJSIP/1000-0000003f", "20") in new stack    

77965   [2022-03-02 14:51:48] VERBOSE[2455] res_pjsip_logger.c: <--- Transmitting SIP response (583 bytes) to UDP:213.57.xx.xx:53516 --->  

77966   SIP/2.0 503 Service Unavailable

77967   Via: SIP/2.0/UDP 192.168.1.21:53516;rport=53516;received=213.57.xx.xx;branch=z9hG4bK-524287-1---cff44688f60fbc60    

77968   Call-ID: Hu46S0_cjNe8yJizUhBxYw..  

77969   From: <sip:[email protected]>;tag=e2092367

77970   To: <sip:[email protected]>;tag=663217ec-16a9-41eb-9f6f-7756f5f17ef7  

77971   CSeq: 2 INVITE  

77972   Server: FPBX-16.0.15(16.20.0)  

77973   Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER  

77974   Reason: Q.850;cause=34  

77975   P-Asserted-Identity: "CID:1000" <sip:[email protected]>    

77976   Content-Length: 0  

77977      

77978      

77979   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] app_macro.c: Spawn extension (macro-dialout-trunk, s-NOANSWER, 4) exited non-zero on 'PJSIP/1000-0000003f' in macro 'dialout-trunk'

77980   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx.c: Spawn extension (from-internal, 966920009722, 12) exited non-zero on 'PJSIP/1000-0000003f'  

77981   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:1] Macro("PJSIP/1000-0000003f", "hangupcall") in new stack    

77982   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:1] GotoIf("PJSIP/1000-0000003f", "1?theend") in new stack  

77983   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx_builtins.c: Goto (macro-hangupcall,s,3)

77984   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:3] ExecIf("PJSIP/1000-0000003f", "0?Set(CDR(recordingfile)=)") in new stack    

77985   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:4] NoOp("PJSIP/1000-0000003f", " montior file= /var/spool/asterisk/monitor/2022/03/02/out-966920009722-1000-20220302-145139-1646232699.63.wav") in new stack  

77986   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:5] GotoIf("PJSIP/1000-0000003f", "1?skipagi") in new stack

77987   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx_builtins.c: Goto (macro-hangupcall,s,7)

77988   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:7] Hangup("PJSIP/1000-0000003f", "") in new stack  

77989   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on 'PJSIP/1000-0000003f' in macro 'hangupcall'    

77990   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/1000-0000003f'  

77991   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] app_stack.c: PJSIP/1000-0000003f Internal Gosub(crm-hangup,s,1) start  

77992   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:1] NoOp("PJSIP/1000-0000003f", "Sending Hangup to CRM") in new stack

77993   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:2] NoOp("PJSIP/1000-0000003f", "HANGUP CAUSE: 34") in new stack  

77994   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:3] ExecIf("PJSIP/1000-0000003f", "0?Set(__CRM_VOICEMAIL=)") in new stack

77995   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:4] NoOp("PJSIP/1000-0000003f", "MASTER CHANNEL: 1646232699.63 = 1646232699.63") in new stack

77996   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:5] GotoIf("PJSIP/1000-0000003f", "0?return") in new stack    

77997   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:6] Set("PJSIP/1000-0000003f", "__CRM_HANGUP=1") in new stack

77998   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:7] AGI("PJSIP/1000-0000003f", "agi://127.0.0.1/sangomacrm.agi") in new stack

77999   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] res_agi.c: <PJSIP/1000-0000003f>AGI Script agi://127.0.0.1/sangomacrm.agi completed, returning 0    

78000   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] pbx.c: Executing [[email protected]:8] Return("PJSIP/1000-0000003f", "") in new stack    

78001   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/1000-0000003f'

78002   [2022-03-02 14:51:48] VERBOSE[3382][C-00000021] app_stack.c: PJSIP/1000-0000003f Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=  

78003   [2022-03-02 14:51:48] VERBOSE[3383][C-00000021] app_mixmonitor.c: MixMonitor close filestream (mixed)  

78004   [2022-03-02 14:51:48] VERBOSE[3383][C-00000021] app_mixmonitor.c: End MixMonitor Recording PJSIP/1000-0000003f  

78005   [2022-03-02 14:51:49] VERBOSE[2242] res_pjsip_logger.c: <--- Received SIP request (375 bytes) from UDP:213.57.179.91:53516 --->

78006   ACK sip:[email protected];transport=UDP SIP/2.0    

78007   Via: SIP/2.0/UDP 192.168.1.21:53516;branch=z9hG4bK-524287-1---cff44688f60fbc60;rport    

78008   Max-Forwards: 70    

78009   To: <sip:[email protected]>;tag=663217ec-16a9-41eb-9f6f-7756f5f17ef7  

78010   From: <sip:[email protected];transport=UDP>;tag=e2092367  

78011   Call-ID: Hu46S0_cjNe8yJizUhBxYw..  

78012   CSeq: 2 ACK

78013   Content-Length: 0  

78014      

78015      

78016   [2022-03-02 14:51:53] VERBOSE[2242] res_pjsip_logger.c: <--- Received SIP request (878 bytes) from UDP:213.57.xx.xx:53516 --->  

78017   REGISTER sip:pbx.sipxxp.xyz;transport=UDP SIP/2.0  

78018   Via: SIP/2.0/UDP 192.168.1.21:53516;branch=z9hG4bK-524287-1---b3f532aa8b00bd2e;rport    

78019   Max-Forwards: 70    

78020   Contact: <sip:[email protected]:53516;transport=UDP;rinstance=5eccecac6d08c98c>

78021   To: <sip:[email protected];transport=UDP>  

78022   From: <sip:[email protected];transport=UDP>;tag=f230b964  

78023   Call-ID: joE7ALTRh8Hn1MxWvim0ow..  

78024   CSeq: 13 REGISTER  

78025   Expires: 60

78026   Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE  

78027   User-Agent: Zoiper v2.10.17.3  

78028   Authorization: Digest username="1000",realm="asterisk",nonce="1646232697/ff952cb0fcc9bcb28b7266c6319c2230",uri="sip:pbx.sipxx.xyz;transport=UDP",response="0222a7927d79c6b4dcb9a0059f74e201",cnonce="a9579549456207d4c66bee81edb4b2ae",nc=00000002,qop=auth,algorithm=md5,opaque="71b569f45b0dbed4"

78029   Allow-Events: presence, kpml, talk  

78030   Content-Length: 0  

78031      

78032      

78033   [2022-03-02 14:51:53] VERBOSE[3256] res_pjsip_logger.c: <--- Transmitting SIP response (486 bytes) to UDP:213.57.xx.xx:53516 --->  

78034   SIP/2.0 200 OK  

78035   Via: SIP/2.0/UDP 192.168.1.21:53516;rport=53516;received=213.57.xx.xx;branch=z9hG4bK-524287-1---b3f532aa8b00bd2e    

78036   Call-ID: joE7ALTRh8Hn1MxWvim0ow..  

78037   From: <sip:[email protected]>;tag=f230b964

78038   To: <sip:[email protected]>;tag=z9hG4bK-524287-1---b3f532aa8b00bd2e    

78039   CSeq: 13 REGISTER  

78040   Date: Wed, 02 Mar 2022 14:51:53 GMT

78041   Contact: <sip:[email protected]:53516;rinstance=5eccecac6d08c98c>;expires=59    

78042   Expires: 60

78043   Server: FPBX-16.0.15(16.20.0)  

78044   Content-Length: 0  

78045      

78046      

78047   [2022-03-02 14:52:24] VERBOSE[3256] res_pjsip_logger.c: <--- Transmitting SIP request (449 bytes) to UDP:51.211.xx.xx:5060 --->

78048   OPTIONS sip:[email protected]:5060 SIP/2.0    

78049   Via: SIP/2.0/UDP 51.211.xx.xx:5060;rport;branch=z9hG4bKPj8907f40a-d7d5-43f3-bce3-e1bb9e15c3c8  

78050   From: <sip:[email protected]>;tag=88ba183d-1004-4461-b4e5-f6ce65ebe0c2    

78051   To: <sip:[email protected]>  

78052   Contact: <sip:[email protected]:5060>  

78053   Call-ID: 14215bba-6ff8-4bb6-ad44-f733c9d6332b  

78054   CSeq: 37106 OPTIONS

78055   Max-Forwards: 70    

78056   User-Agent: FPBX-16.0.15(16.20.0)  

78057   Content-Length: 0  

78058      

78059      

78060   [2022-03-02 14:52:24] VERBOSE[2242] res_pjsip_logger.c: <--- Received SIP response (520 bytes) from UDP:51.211.xx.xx:5060 --->  

78061   SIP/2.0 200 OK  

78062   Via: SIP/2.0/UDP 51.211.xx.xx:5060;branch=z9hG4bKPj8907f40a-d7d5-43f3-bce3-e1bb9e15c3c8;received=70.34.xx.xx;rport=5060

78063   From: <sip:[email protected]>;tag=88ba183d-1004-4461-b4e5-f6ce65ebe0c2    

78064   To: <sip:[email protected]>;tag=as44fd7888    

78065   Call-ID: 14215bba-6ff8-4bb6-ad44-f733c9d6332b  

78066   CSeq: 37106 OPTIONS

78067   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO    

78068   Supported: replaces, timer  

78069   Contact: <sip:192.168.100.69>  

78070   Accept: application/sdp

78071   Content-Length: 0  

78072      

78073      

78074   [2022-03-02 14:52:33] VERBOSE[2242] res_pjsip_logger.c: <--- Received SIP request (494 bytes) from UDP:51.211.xx.xx:5060 --->  

78075   OPTIONS sip:[email protected]:5060 SIP/2.0

78076   Via: SIP/2.0/UDP 192.168.100.69:5060;branch=z9hG4bK7032f84c;rport  

78077   Max-Forwards: 70    

78078   From: "Unknown" <sip:[email protected]>;tag=as15a3e4af

78079   To: <sip:[email protected]:5060>  

78080   Contact: <sip:[email protected]>  

78081   Call-ID: [email protected]    

78082   CSeq: 102 OPTIONS  

78083   Date: Wed, 02 Mar 2022 14:52:32 GMT

78084   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO    

78085   Supported: replaces, timer  

78086   Content-Length: 0  

78087      

78088      

78089   [2022-03-02 14:52:33] VERBOSE[3256] res_pjsip_logger.c: <--- Transmitting SIP response (832 bytes) to UDP:51.211.xx.xx:5060 --->    

78090   SIP/2.0 200 OK  

78091   Via: SIP/2.0/UDP 192.168.100.69:5060;rport=5060;received=51.211.xx.xx;branch=z9hG4bK7032f84c    

78092   Call-ID: [email protected]    

78093   From: "Unknown" <sip:[email protected]>;tag=as15a3e4af

78094   To: <sip:[email protected]>;tag=z9hG4bK7032f84c    

78095   CSeq: 102 OPTIONS  

78096   Accept: application/sdp, application/dialog-info+xml, application/xpidf+xml, application/cpim-pidf+xml, application/simple-message-summary, application/pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, message/sipfrag;version=2.0    

78097   Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER  

78098   Supported: 100rel, timer, replaces, norefersub  

78099   Accept-Encoding: identity  

78100   Accept-Language: en

78101   Server: FPBX-16.0.15(16.20.0)  

78102   Content-Length: 0  

78103      

78104      

78105   [2022-03-02 14:52:38] VERBOSE[2455] res_pjsip_logger.c: <--- Transmitting SIP request (480 bytes) to UDP:213.57.xx.xx:53516 --->    

78106   OPTIONS sip:[email protected]:53516;rinstance=5eccecac6d08c98c SIP/2.0  

78107   Via: SIP/2.0/UDP 51.211.xx.xx:5060;rport;branch=z9hG4bKPja62fd34c-91fd-4322-b2da-d055647cd884  

78108   From: <sip:[email protected]>;tag=c7798b7f-9e86-4db8-8864-a65f6e8dfcef  

78109   To: <sip:[email protected];rinstance=5eccecac6d08c98c>  

78110   Contact: <sip:[email protected]:5060>  

78111   Call-ID: f32ce780-5af8-4ab3-9cb5-7a22d5e27f74  

78112   CSeq: 23480 OPTIONS

78113   Max-Forwards: 70    

78114   User-Agent: FPBX-16.0.15(16.20.0)  

78115   Content-Length: 0  

78116      

78117      

78118   [2022-03-02 14:52:38] VERBOSE[2242] res_pjsip_logger.c: <--- Transmitting SIP request (480 bytes) to UDP:213.57.xx.xx:53516 --->    

78119   OPTIONS sip:[email protected]:53516;rinstance=5eccecac6d08c98c SIP/2.0  

78120   Via: SIP/2.0/UDP 51.211.xx.xx:5060;rport;branch=z9hG4bKPja62fd34c-91fd-4322-b2da-d055647cd884  

78121   From: <sip:[email protected]>;tag=c7798b7f-9e86-4db8-8864-a65f6e8dfcef  

78122   To: <sip:[email protected];rinstance=5eccecac6d08c98c>  

78123   Contact: <sip:[email protected]:5060>  

78124   Call-ID: f32ce780-5af8-4ab3-9cb5-7a22d5e27f74  

78125   CSeq: 23480 OPTIONS

78126   Max-Forwards: 70    

78127   User-Agent: FPBX-16.0.15(16.20.0)  

78128   Content-Length: 0  

78129      

78130      

78131   [2022-03-02 14:52:39] VERBOSE[2242] res_pjsip_logger.c: <--- Transmitting SIP request (480 bytes) to UDP:213.57.xx.xx:53516 --->    

78132   OPTIONS sip:[email protected]:53516;rinstance=5eccecac6d08c98c SIP/2.0  

78133   Via: SIP/2.0/UDP 51.211.xx.xx:5060;rport;branch=z9hG4bKPja62fd34c-91fd-4322-b2da-d055647cd884  

78134   From: <sip:[email protected]>;tag=c7798b7f-9e86-4db8-8864-a65f6e8dfcef  

78135   To: <sip:[email protected];rinstance=5eccecac6d08c98c>  

78136   Contact: <sip:[email protected]:5060>  

78137   Call-ID: f32ce780-5af8-4ab3-9cb5-7a22d5e27f74  

78138   CSeq: 23480 OPTIONS

78139   Max-Forwards: 70    

78140   User-Agent: FPBX-16.0.15(16.20.0)  

78141   Content-Length: 0  

78142      

78143      

78144   [2022-03-02 14:52:41] VERBOSE[3256] res_pjsip/pjsip_configuration.c: Endpoint 1000 is now Unreachable  

78145   [2022-03-02 14:52:41] VERBOSE[3256] res_pjsip/pjsip_options.c: Contact 1000/sip:[email protected]:53516;rinstance=5eccecac6d08c98c is now Unreachable. RTT: 0.000 msec  

78146   [2022-03-02 14:52:41] VERBOSE[2242] res_pjsip_logger.c: <--- Transmitting SIP request (480 bytes) to UDP:213.57.xx.xx:53516 --->    

78147   OPTIONS sip:[email protected]:53516;rinstance=5eccecac6d08c98c SIP/2.0  

78148   Via: SIP/2.0/UDP 51.211.xx.xx:5060;rport;branch=z9hG4bKPja62fd34c-91fd-4322-b2da-d055647cd884  

78149   From: <sip:[email protected]>;tag=c7798b7f-9e86-4db8-8864-a65f6e8dfcef  

78150   To: <sip:[email protected];rinstance=5eccecac6d08c98c>  

78151   Contact: <sip:[email protected]:5060>  

78152   Call-ID: f32ce780-5af8-4ab3-9cb5-7a22d5e27f74  

78153   CSeq: 23480 OPTIONS

78154   Max-Forwards: 70    

78155   User-Agent: FPBX-16.0.15(16.20.0)  

78156   Content-Length: 0

It would be a very strange system where these would be the same:

It is possible you don’t need an External Address. If you do need one, you generally need Local Networks as well. External address is the address at which your system appears to be seen from the public internet.

My freepbx running in the cloud with no other firewall than system Responsive firewall located in europe and the GSM gateway running on DMZ located in asia and users connect with ext worldwide i tried before without and with the sip server network and ip addr same issue, i get 503 sip unavailable when call disconnect after 10 seconds on silence waiting.

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