No Way Audio - PJSIP 2.10 - Gsm Gateway - Asterisk 18.3.0

Hello Team Good Morning,

There Is Working Scenario

And the 1 Not Working Scenario

Please I need help. For 1 week I have an issue. I know that No Way is very old issue on asterisk but its blocking me since. Like you can see on picture I work with Asterisk on Realtime mode. I created 4 extensions. When I make call between 2 Sip phones all is working fine but when I make call between Sip phone and GSM Gateway, I have no way audio issue. When I capture traffic on device, I see that Sip Phone send RTP to Asterisk but GSM Gateway not send any information to Server and on Debug RTP of Gateway i see @IP 255.255.255.255 not public IP of Asterisk Server.
I tried to setup the GSM Gateway on Asterisk 13 (RealTime) and using SIP and all is working fine. Please can you help me to understand why it work with sip phone but not working with GSM Gateway.

There is my environment details
Asterisk Version: 18.3.0 & 13.28
PJSIP Version: 2.10.0 & 2.4.0
OS Version: Centos 8 & 6
DB Engine: MySQL
DB Connector: Maria-DB-Connector
Hosting Type : Online VPS With Public IP Address

SQL Code FOr ps_aors


– Records of ps_aors


INSERT INTO ps_aors VALUES (‘100’, NULL, NULL, NULL, 2, NULL, NULL, 30, NULL, NULL, NULL, NULL, NULL, NULL);
INSERT INTO ps_aors VALUES (‘101’, NULL, NULL, NULL, 2, NULL, NULL, 30, NULL, NULL, NULL, NULL, NULL, NULL);
INSERT INTO ps_aors VALUES (‘102’, NULL, NULL, NULL, 2, NULL, NULL, 30, NULL, NULL, NULL, NULL, NULL, NULL);
INSERT INTO ps_aors VALUES (‘BOBO_D16’, NULL, NULL, NULL, 16, NULL, NULL, 30, NULL, NULL, NULL, NULL, NULL, NULL);
INSERT INTO ps_aors VALUES (‘IASmartVoIPGw’, NULL, NULL, NULL, 2, NULL, NULL, 30, NULL, NULL, NULL, NULL, NULL, NULL);

SET FOREIGN_KEY_CHECKS = 1;

SQL Code FOr ps_auths


– Records of ps_auths


INSERT INTO ps_auths VALUES (‘100’, ‘userpass’, NULL, NULL, ‘100’, NULL, ‘100’, NULL, NULL, NULL);
INSERT INTO ps_auths VALUES (‘101’, ‘userpass’, NULL, NULL, ‘101’, NULL, ‘101’, NULL, NULL, NULL);
INSERT INTO ps_auths VALUES (‘102’, ‘userpass’, NULL, NULL, ‘102’, NULL, ‘102’, NULL, NULL, NULL);
INSERT INTO ps_auths VALUES (‘BOBO_D16’, ‘userpass’, NULL, NULL, ‘BOBO_D16’, NULL, ‘BOBO_D16’, NULL, NULL, NULL);
INSERT INTO ps_auths VALUES (‘IASmartVoIPGw’, ‘userpass’, NULL, NULL, ‘IASmartVoIPGw’, NULL, ‘IASmartVoIPGw’, NULL, NULL, NULL);

SET FOREIGN_KEY_CHECKS = 1;

SQL Code FOr ps_endpoints


– Records of ps_endpoints


INSERT INTO ps_endpoints VALUES (‘100’, ‘transport-udp’, ‘100’, ‘100’, ‘testing’, ‘all’, ‘all’, ‘no’, NULL, NULL, NULL, ‘yes’, NULL, NULL, ‘yes’, ‘yes’, NULL, ‘100@default’, NULL, NULL, NULL, ‘yes’, NULL, ‘yes’, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, ‘0.0.0.0/0’, ‘0.0.0.0/0’, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL);
INSERT INTO ps_endpoints VALUES (‘101’, ‘transport-udp’, ‘101’, ‘101’, ‘testing’, ‘all’, ‘all’, ‘no’, NULL, NULL, NULL, ‘yes’, NULL, NULL, ‘yes’, ‘yes’, NULL, ‘101@default’, NULL, NULL, NULL, ‘yes’, NULL, ‘yes’, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, ‘0.0.0.0/0’, ‘0.0.0.0/0’, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL);
INSERT INTO ps_endpoints VALUES (‘102’, ‘transport-udp’, ‘102’, ‘102’, ‘testing’, ‘all’, ‘all’, ‘no’, NULL, NULL, NULL, ‘yes’, NULL, NULL, ‘yes’, ‘yes’, NULL, ‘102@default’, NULL, NULL, NULL, ‘yes’, NULL, ‘yes’, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, ‘0.0.0.0/0’, ‘0.0.0.0/0’, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL);
INSERT INTO ps_endpoints VALUES (‘BOBO_D16’, ‘transport-udp’, ‘BOBO_D16’, ‘BOBO_D16’, ‘testing’, ‘all’, ‘all’, ‘no’, NULL, NULL, NULL, NULL, ‘auto’, NULL, ‘yes’, ‘yes’, NULL, ‘BOBO_D16@default’, NULL, NULL, NULL, ‘yes’, NULL, ‘yes’, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, ‘185.189.151.222’, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, ‘yes’, NULL, NULL, ‘0.0.0.0/0’, ‘0.0.0.0/0’, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL);
INSERT INTO ps_endpoints VALUES (‘IASmartVoIPGw’, ‘transport-udp’, ‘IASmartVoIPGw’, ‘IASmartVoIPGw’, ‘testing’, ‘all’, ‘all’, ‘no’, NULL, NULL, NULL, NULL, NULL, NULL, ‘yes’, NULL, NULL, ‘IASmartVoIPGw@default’, NULL, NULL, NULL, ‘yes’, NULL, ‘yes’, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, ‘0.0.0.0/0’, ‘0.0.0.0/0’, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL);

SET FOREIGN_KEY_CHECKS = 1;

Please realy need help

Continuing the discussion from No Way Audio - PJSIP 2.10 - Gsm Gateway - Asterisk 18.3.0:

There is image of RTP of GSM Gateway

image

Like you can see i the device is dont send data to Good IP of server but it’s registred fine
Like you can see there

Please team need help

Thank’s in advance

There is no team. This is a peer support (user to user) forum.

Please provide the pjsip set debug on on output for the INVITE to the gateway, all it’s responses, and the ACK that closes the transaction. Please provide the same information captured at the gateway, if it differs.

Sorry for mistake

There is pjsip set logger on




<--- Received SIP request (1197 bytes) from UDP:154.72.150.215:42675 --->
INVITE sip:237659856358@185.189.151.222 SIP/2.0
Via: SIP/2.0/UDP 10.153.1.160:10197;branch=z9hG4bK-524287-1---f370142b10665922;rport
Max-Forwards: 70
Contact: <sip:102@154.72.150.215:42675>;+sip.instance="<urn:uuid:e20f347c-4e9e-46a3-a10d-272460158a9b>"
To: <sip:237659856358@185.189.151.222>
From: <sip:102@185.189.151.222>;tag=04346b00
Call-ID: 3XcPn6ApEkMfoXwz1aqwIg..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH
Content-Type: application/sdp
Supported: replaces, answermode, eventlist, outbound, path
User-Agent: PortSIP UC Client
Allow-Events: hold, talk, conference, dialog
Content-Length: 509

v=0
o=- 1618146237 1 IN IP4 10.153.1.160
s=ps
c=IN IP4 10.153.1.160
t=0 0
m=audio 10016 RTP/AVP 18 8 0 3 9 97 98 99 100 102 105 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 AMR/8000
a=rtpmap:99 AMR-WB/16000
a=rtpmap:100 SPEEX/8000
a=rtpmap:102 SPEEX/16000
a=rtpmap:105 opus/48000/2
a=fmtp:105 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

<--- Transmitting SIP response (516 bytes) to UDP:154.72.150.215:42675 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.153.1.160:10197;rport=42675;received=154.72.150.215;branch=z9hG4bK-524287-1---f370142b10665922
Call-ID: 3XcPn6ApEkMfoXwz1aqwIg..
From: <sip:102@185.189.151.222>;tag=04346b00
To: <sip:237659856358@185.189.151.222>;tag=z9hG4bK-524287-1---f370142b10665922
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1618146237/474635e7f656d1f49c4bf7c494221305",opaque="2c58992b56844203",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.3.0
Content-Length:  0


<--- Received SIP request (345 bytes) from UDP:154.72.150.215:42675 --->
ACK sip:237659856358@185.189.151.222 SIP/2.0
Via: SIP/2.0/UDP 10.153.1.160:10197;branch=z9hG4bK-524287-1---f370142b10665922;rport
Max-Forwards: 70
To: <sip:237659856358@185.189.151.222>;tag=z9hG4bK-524287-1---f370142b10665922
From: <sip:102@185.189.151.222>;tag=04346b00
Call-ID: 3XcPn6ApEkMfoXwz1aqwIg..
CSeq: 1 ACK
Content-Length: 0


<--- Received SIP request (1490 bytes) from UDP:154.72.150.215:42675 --->
INVITE sip:237659856358@185.189.151.222 SIP/2.0
Via: SIP/2.0/UDP 10.153.1.160:10197;branch=z9hG4bK-524287-1---afc2fb4dd695a45e;rport
Max-Forwards: 70
Contact: <sip:102@154.72.150.215:42675>;+sip.instance="<urn:uuid:e20f347c-4e9e-46a3-a10d-272460158a9b>"
To: <sip:237659856358@185.189.151.222>
From: <sip:102@185.189.151.222>;tag=04346b00
Call-ID: 3XcPn6ApEkMfoXwz1aqwIg..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH
Content-Type: application/sdp
Supported: replaces, answermode, eventlist, outbound, path
User-Agent: PortSIP UC Client
Authorization: Digest username="102",realm="asterisk",nonce="1618146237/474635e7f656d1f49c4bf7c494221305",uri="sip:237659856358@185.189.151.222",response="0a02a41a6be9061934f38ad777b8f7ce",cnonce="7171a5d85957755e916091ba49406c55",nc=00000001,qop=auth,algorithm=md5,opaque="2c58992b56844203"
Allow-Events: hold, talk, conference, dialog
Content-Length: 509

v=0
o=- 1618146237 1 IN IP4 10.153.1.160
s=ps
c=IN IP4 10.153.1.160
t=0 0
m=audio 10016 RTP/AVP 18 8 0 3 9 97 98 99 100 102 105 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 AMR/8000
a=rtpmap:99 AMR-WB/16000
a=rtpmap:100 SPEEX/8000
a=rtpmap:102 SPEEX/16000
a=rtpmap:105 opus/48000/2
a=fmtp:105 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

<--- Transmitting SIP response (324 bytes) to UDP:154.72.150.215:42675 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.153.1.160:10197;rport=42675;received=154.72.150.215;branch=z9hG4bK-524287-1---afc2fb4dd695a45e
Call-ID: 3XcPn6ApEkMfoXwz1aqwIg..
From: <sip:102@185.189.151.222>;tag=04346b00
To: <sip:237659856358@185.189.151.222>
CSeq: 2 INVITE
Server: Asterisk PBX 18.3.0
Content-Length:  0


    -- Executing [237659856358@testing:1] NoOp("PJSIP/102-00000000", "") in new stack
    -- Executing [237659856358@testing:2] Dial("PJSIP/102-00000000", "PJSIP/237659856358@BOBO_D16") in new stack
    -- Called PJSIP/237659856358@BOBO_D16
<--- Transmitting SIP request (650 bytes) to UDP:129.0.205.28:49101 --->
INVITE sip:237659856358@129.0.205.28:49101 SIP/2.0
Via: SIP/2.0/UDP 185.189.151.222:6060;rport;branch=z9hG4bKPj7f01f1da-016c-44c8-80e6-476254c6ec67
From: <sip:102@185.189.151.222>;tag=7735c908-241a-4b16-b172-e8b80fdf9df2
To: <sip:237659856358@129.0.205.28>
Contact: <sip:asterisk@185.189.151.222:6060>
Call-ID: 3575c280-e110-4ca9-8534-6cb6a9226bde
CSeq: 7049 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Length:  0


<--- Received SIP request (540 bytes) from UDP:129.0.205.28:49101 --->
REGISTER sip:185.189.151.222:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.115:6060;branch=z9hG4bKd06f46b36be41e1aa58089db24ce8d4b;rport
From: <sip:BOBO_D16@185.189.151.222:6060>;tag=71a259c2597821e914db9b56bcea8d04
To: <sip:BOBO_D16@185.189.151.222:6060>
Call-ID: 611e5f6241949164b839b1e7587988de@192.168.101.115
CSeq: 1846984758 REGISTER
Contact: <sip:BOBO_D16@192.168.101.115:6060>
Expires: 6
User-Agent: AIO-VF  01231410
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
Max-Forwards: 70
Content-Length: 0


<--- Transmitting SIP response (584 bytes) to UDP:129.0.205.28:49101 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.101.115:6060;rport=49101;received=129.0.205.28;branch=z9hG4bKd06f46b36be41e1aa58089db24ce8d4b
Call-ID: 611e5f6241949164b839b1e7587988de@192.168.101.115
From: <sip:BOBO_D16@185.189.151.222>;tag=71a259c2597821e914db9b56bcea8d04
To: <sip:BOBO_D16@185.189.151.222>;tag=z9hG4bKd06f46b36be41e1aa58089db24ce8d4b
CSeq: 1846984758 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1618146238/29ef8a937e92da46f5a1aee04b94dd26",opaque="0fcf389f34999774",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.3.0
Content-Length:  0


<--- Received SIP response (321 bytes) from UDP:129.0.205.28:49101 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 185.189.151.222:6060;rport=6060;branch=z9hG4bKPj7f01f1da-016c-44c8-80e6-476254c6ec67
From: <sip:102@185.189.151.222>;tag=7735c908-241a-4b16-b172-e8b80fdf9df2
To: <sip:237659856358@129.0.205.28>
Call-ID: 3575c280-e110-4ca9-8534-6cb6a9226bde
CSeq: 7049 INVITE
Content-Length: 0


<--- Received SIP request (839 bytes) from UDP:129.0.205.28:49101 --->
REGISTER sip:185.189.151.222:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.115:6060;branch=z9hG4bKac0b91252cbc04975de54a87141348b0;rport
From: <sip:BOBO_D16@185.189.151.222:6060>;tag=71a259c2597821e914db9b56bcea8d04
To: <sip:BOBO_D16@185.189.151.222:6060>
Call-ID: 611e5f6241949164b839b1e7587988de@192.168.101.115
CSeq: 1846984759 REGISTER
Contact: <sip:BOBO_D16@192.168.101.115:6060>
Authorization: Digest username="BOBO_D16", realm="asterisk", nonce="1618146238/29ef8a937e92da46f5a1aee04b94dd26", uri="sip:185.189.151.222:6060", response="ea40558db605914d790513cab7fbc845", algorithm=md5, cnonce="d73f64c2d32b87aaacb6156619189e62", opaque="0fcf389f34999774", qop=auth, nc=00000001
Expires: 6
User-Agent: AIO-VF  01231410
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
Max-Forwards: 70
Content-Length: 0


<--- Transmitting SIP request (657 bytes) to UDP:129.0.205.28:49101 --->
NOTIFY sip:BOBO_D16@129.0.205.28:49101 SIP/2.0
Via: SIP/2.0/UDP 185.189.151.222:6060;rport;branch=z9hG4bKPj4f9edd82-fab6-474a-a9e1-e4243d954338
From: <sip:BOBO_D16@185.189.151.222>;tag=bee12a28-be7d-4e76-8a9e-173760d230e0
To: <sip:BOBO_D16@129.0.205.28>
Contact: <sip:BOBO_D16@185.189.151.222:6060>
Call-ID: 15030e77-dc16-4112-994a-b98d7ddb94f4
CSeq: 59958 NOTIFY
Subscription-State: terminated
Event: message-summary
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Type: application/simple-message-summary
Content-Length:    48

Messages-Waiting: no
Voice-Message: 0/0 (0/0)

<--- Transmitting SIP response (535 bytes) to UDP:129.0.205.28:49101 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.101.115:6060;rport=49101;received=129.0.205.28;branch=z9hG4bKac0b91252cbc04975de54a87141348b0
Call-ID: 611e5f6241949164b839b1e7587988de@192.168.101.115
From: <sip:BOBO_D16@185.189.151.222>;tag=71a259c2597821e914db9b56bcea8d04
To: <sip:BOBO_D16@185.189.151.222>;tag=z9hG4bKac0b91252cbc04975de54a87141348b0
CSeq: 1846984759 REGISTER
Date: Sun, 11 Apr 2021 13:03:58 GMT
Contact: <sip:BOBO_D16@192.168.101.115:6060>;expires=59
Expires: 60
Server: Asterisk PBX 18.3.0
Content-Length:  0


<--- Received SIP response (356 bytes) from UDP:129.0.205.28:49101 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 185.189.151.222:6060;rport=6060;branch=z9hG4bKPj4f9edd82-fab6-474a-a9e1-e4243d954338
From: <sip:BOBO_D16@185.189.151.222>;tag=bee12a28-be7d-4e76-8a9e-173760d230e0
To: <sip:BOBO_D16@129.0.205.28>;tag=ac9870ab67f0089589a3b3a83e98dba9
Call-ID: 15030e77-dc16-4112-994a-b98d7ddb94f4
CSeq: 59958 NOTIFY
Content-Length: 0


<--- Received SIP request (612 bytes) from UDP:129.0.205.28:49101 --->
SUBSCRIBE sip:BOBO_D16@185.189.151.222:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.115:6060;branch=z9hG4bK98f355ccd32ba80d8bd740e5c8e291c0;rport
From: <sip:BOBO_D16@185.189.151.222:6060>;tag=a7af52d79fcdcc32b807389c2fc99cea
To: <sip:BOBO_D16@185.189.151.222:6060>
Call-ID: cd41db78275089f560524266086ca4d5@192.168.101.115
CSeq: 1846984943 SUBSCRIBE
Contact: <sip:BOBO_D16@192.168.101.115:6060>
Event: message-summary
Supported: eventlist
Expires: 16
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
Max-Forwards: 70
Accept: application/simple-message-summary
Content-Length: 0


<--- Transmitting SIP response (585 bytes) to UDP:129.0.205.28:49101 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.101.115:6060;rport=49101;received=129.0.205.28;branch=z9hG4bK98f355ccd32ba80d8bd740e5c8e291c0
Call-ID: cd41db78275089f560524266086ca4d5@192.168.101.115
From: <sip:BOBO_D16@185.189.151.222>;tag=a7af52d79fcdcc32b807389c2fc99cea
To: <sip:BOBO_D16@185.189.151.222>;tag=z9hG4bK98f355ccd32ba80d8bd740e5c8e291c0
CSeq: 1846984943 SUBSCRIBE
WWW-Authenticate: Digest realm="asterisk",nonce="1618146238/29ef8a937e92da46f5a1aee04b94dd26",opaque="5d2cc95464021e15",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.3.0
Content-Length:  0


<--- Received SIP request (920 bytes) from UDP:129.0.205.28:49101 --->
SUBSCRIBE sip:BOBO_D16@185.189.151.222:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.115:6060;branch=z9hG4bKf496dc06e55e18a3d9b76f3aaf2f3996;rport
From: <sip:BOBO_D16@185.189.151.222:6060>;tag=a7af52d79fcdcc32b807389c2fc99cea
To: <sip:BOBO_D16@185.189.151.222:6060>
Call-ID: cd41db78275089f560524266086ca4d5@192.168.101.115
CSeq: 1846984944 SUBSCRIBE
Contact: <sip:BOBO_D16@192.168.101.115:6060>
Authorization: Digest username="BOBO_D16", realm="asterisk", nonce="1618146238/29ef8a937e92da46f5a1aee04b94dd26", uri="sip:BOBO_D16@185.189.151.222:6060", response="8d4bbcdb9101b632daaf3af254f0cbcd", algorithm=md5, cnonce="86a8a1653ac705aba42bbc6de0b221d1", opaque="5d2cc95464021e15", qop=auth, nc=00000001
Event: message-summary
Supported: eventlist
Expires: 16
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
Max-Forwards: 70
Accept: application/simple-message-summary
Content-Length: 0


<--- Transmitting SIP response (436 bytes) to UDP:129.0.205.28:49101 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.101.115:6060;rport=49101;received=129.0.205.28;branch=z9hG4bKf496dc06e55e18a3d9b76f3aaf2f3996
Call-ID: cd41db78275089f560524266086ca4d5@192.168.101.115
From: <sip:BOBO_D16@185.189.151.222>;tag=a7af52d79fcdcc32b807389c2fc99cea
To: <sip:BOBO_D16@185.189.151.222>;tag=z9hG4bKf496dc06e55e18a3d9b76f3aaf2f3996
CSeq: 1846984944 SUBSCRIBE
Server: Asterisk PBX 18.3.0
Content-Length:  0


<--- Transmitting SIP request (446 bytes) to UDP:129.0.205.28:49101 --->
OPTIONS sip:BOBO_D16@129.0.205.28:49101 SIP/2.0
Via: SIP/2.0/UDP 185.189.151.222:6060;rport;branch=z9hG4bKPjef79be6f-d1b4-43da-b481-2968fd9bb8c8
From: <sip:BOBO_D16@185.189.151.222>;tag=e69d1479-dd1b-48f0-a741-94877b576cf0
To: <sip:BOBO_D16@129.0.205.28>
Contact: <sip:BOBO_D16@185.189.151.222:6060>
Call-ID: e49953be-ccb1-42ba-9191-f9f2200401c9
CSeq: 57677 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Length:  0


<--- Received SIP response (495 bytes) from UDP:129.0.205.28:49101 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 185.189.151.222:6060;rport=6060;branch=z9hG4bKPjef79be6f-d1b4-43da-b481-2968fd9bb8c8
From: <sip:BOBO_D16@185.189.151.222>;tag=e69d1479-dd1b-48f0-a741-94877b576cf0
To: <sip:BOBO_D16@129.0.205.28>;tag=39947cbb243c9fca0cbb0441333aab3f
Call-ID: e49953be-ccb1-42ba-9191-f9f2200401c9
CSeq: 57677 OPTIONS
Contact: <sip:BOBO_D16@192.168.101.115:6060>
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
Content-Length: 0


<--- Received SIP request (612 bytes) from UDP:129.0.205.28:49101 --->
SUBSCRIBE sip:BOBO_D16@185.189.151.222:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.115:6060;branch=z9hG4bK925119710204a64fc80d861c01d9144c;rport
From: <sip:BOBO_D16@185.189.151.222:6060>;tag=c5036d879950a94492de2c1e59b2236c
To: <sip:BOBO_D16@185.189.151.222:6060>
Call-ID: 090b2e505547f4574a5fd436285e49f3@192.168.101.115
CSeq: 1846984944 SUBSCRIBE
Contact: <sip:BOBO_D16@192.168.101.115:6060>
Event: message-summary
Supported: eventlist
Expires: 16
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
Max-Forwards: 70
Accept: application/simple-message-summary
Content-Length: 0


<--- Transmitting SIP response (585 bytes) to UDP:129.0.205.28:49101 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.101.115:6060;rport=49101;received=129.0.205.28;branch=z9hG4bK925119710204a64fc80d861c01d9144c
Call-ID: 090b2e505547f4574a5fd436285e49f3@192.168.101.115
From: <sip:BOBO_D16@185.189.151.222>;tag=c5036d879950a94492de2c1e59b2236c
To: <sip:BOBO_D16@185.189.151.222>;tag=z9hG4bK925119710204a64fc80d861c01d9144c
CSeq: 1846984944 SUBSCRIBE
WWW-Authenticate: Digest realm="asterisk",nonce="1618146248/f5259d97db3b018f38abc1140db74b61",opaque="1e8f062d4782595b",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.3.0
Content-Length:  0


<--- Received SIP request (920 bytes) from UDP:129.0.205.28:49101 --->
SUBSCRIBE sip:BOBO_D16@185.189.151.222:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.115:6060;branch=z9hG4bKbc7ff3989335f7ed35a2469ca1f9a6d5;rport
From: <sip:BOBO_D16@185.189.151.222:6060>;tag=c5036d879950a94492de2c1e59b2236c
To: <sip:BOBO_D16@185.189.151.222:6060>
Call-ID: 090b2e505547f4574a5fd436285e49f3@192.168.101.115
CSeq: 1846984945 SUBSCRIBE
Contact: <sip:BOBO_D16@192.168.101.115:6060>
Authorization: Digest username="BOBO_D16", realm="asterisk", nonce="1618146248/f5259d97db3b018f38abc1140db74b61", uri="sip:BOBO_D16@185.189.151.222:6060", response="81b827514c999139351af1e86164090a", algorithm=md5, cnonce="e69fdb1dd7fb4d0f2049c63a48ae0004", opaque="1e8f062d4782595b", qop=auth, nc=00000001
Event: message-summary
Supported: eventlist
Expires: 16
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
Max-Forwards: 70
Accept: application/simple-message-summary
Content-Length: 0


<--- Transmitting SIP response (436 bytes) to UDP:129.0.205.28:49101 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.101.115:6060;rport=49101;received=129.0.205.28;branch=z9hG4bKbc7ff3989335f7ed35a2469ca1f9a6d5
Call-ID: 090b2e505547f4574a5fd436285e49f3@192.168.101.115
From: <sip:BOBO_D16@185.189.151.222>;tag=c5036d879950a94492de2c1e59b2236c
To: <sip:BOBO_D16@185.189.151.222>;tag=z9hG4bKbc7ff3989335f7ed35a2469ca1f9a6d5
CSeq: 1846984945 SUBSCRIBE
Server: Asterisk PBX 18.3.0
Content-Length:  0


<--- Transmitting SIP request (430 bytes) to UDP:154.72.150.215:42675 --->
OPTIONS sip:102@154.72.150.215:42675 SIP/2.0
Via: SIP/2.0/UDP 185.189.151.222:6060;rport;branch=z9hG4bKPj79a887ea-0b58-403e-950d-543333edfe72
From: <sip:102@185.189.151.222>;tag=6f36af5d-7bf3-4729-8d30-2bb8295daa29
To: <sip:102@154.72.150.215>
Contact: <sip:102@185.189.151.222:6060>
Call-ID: 721c7fa6-346f-44ff-9416-26e6d20dc61c
CSeq: 24896 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Length:  0


<--- Transmitting SIP request (430 bytes) to UDP:154.72.150.215:42675 --->
OPTIONS sip:102@154.72.150.215:42675 SIP/2.0
Via: SIP/2.0/UDP 185.189.151.222:6060;rport;branch=z9hG4bKPj79a887ea-0b58-403e-950d-543333edfe72
From: <sip:102@185.189.151.222>;tag=6f36af5d-7bf3-4729-8d30-2bb8295daa29
To: <sip:102@154.72.150.215>
Contact: <sip:102@185.189.151.222:6060>
Call-ID: 721c7fa6-346f-44ff-9416-26e6d20dc61c
CSeq: 24896 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Length:  0


<--- Transmitting SIP request (430 bytes) to UDP:154.72.150.215:42675 --->
OPTIONS sip:102@154.72.150.215:42675 SIP/2.0
Via: SIP/2.0/UDP 185.189.151.222:6060;rport;branch=z9hG4bKPj79a887ea-0b58-403e-950d-543333edfe72
From: <sip:102@185.189.151.222>;tag=6f36af5d-7bf3-4729-8d30-2bb8295daa29
To: <sip:102@154.72.150.215>
Contact: <sip:102@185.189.151.222:6060>
Call-ID: 721c7fa6-346f-44ff-9416-26e6d20dc61c
CSeq: 24896 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Length:  0


<--- Received SIP response (711 bytes) from UDP:154.72.150.215:42675 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 185.189.151.222:6060;rport=6060;branch=z9hG4bKPj79a887ea-0b58-403e-950d-543333edfe72
Contact: <sip:10.153.1.160:10197>
To: <sip:102@154.72.150.215>;tag=140b9f3e
From: <sip:102@185.189.151.222>;tag=6f36af5d-7bf3-4729-8d30-2bb8295daa29
Call-ID: 721c7fa6-346f-44ff-9416-26e6d20dc61c
CSeq: 24896 OPTIONS
Accept: application/sdp, multipart/mixed, multipart/signed, multipart/alternative, application/vnd.3gpp.cw+xml
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH
Supported: replaces, answermode, eventlist, outbound, path
User-Agent: PortSIP UC Client
Allow-Events: hold, talk, conference, dialog
Content-Length: 0


<--- Received SIP response (711 bytes) from UDP:154.72.150.215:42675 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 185.189.151.222:6060;rport=6060;branch=z9hG4bKPj79a887ea-0b58-403e-950d-543333edfe72
Contact: <sip:10.153.1.160:10197>
To: <sip:102@154.72.150.215>;tag=140b9f3e
From: <sip:102@185.189.151.222>;tag=6f36af5d-7bf3-4729-8d30-2bb8295daa29
Call-ID: 721c7fa6-346f-44ff-9416-26e6d20dc61c
CSeq: 24896 OPTIONS
Accept: application/sdp, multipart/mixed, multipart/signed, multipart/alternative, application/vnd.3gpp.cw+xml
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH
Supported: replaces, answermode, eventlist, outbound, path
User-Agent: PortSIP UC Client
Allow-Events: hold, talk, conference, dialog
Content-Length: 0


<--- Received SIP request (612 bytes) from UDP:129.0.205.28:49101 --->
SUBSCRIBE sip:BOBO_D16@185.189.151.222:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.115:6060;branch=z9hG4bK4520303178ac55aaa7db5c740b3f635c;rport
From: <sip:BOBO_D16@185.189.151.222:6060>;tag=260b305c6f7b37c807cb5e3927b9582a
To: <sip:BOBO_D16@185.189.151.222:6060>
Call-ID: 39f9e932d93990fcb552f50ea302923f@192.168.101.115
CSeq: 1846984945 SUBSCRIBE
Contact: <sip:BOBO_D16@192.168.101.115:6060>
Event: message-summary
Supported: eventlist
Expires: 16
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
Max-Forwards: 70
Accept: application/simple-message-summary
Content-Length: 0


<--- Transmitting SIP response (585 bytes) to UDP:129.0.205.28:49101 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.101.115:6060;rport=49101;received=129.0.205.28;branch=z9hG4bK4520303178ac55aaa7db5c740b3f635c
Call-ID: 39f9e932d93990fcb552f50ea302923f@192.168.101.115
From: <sip:BOBO_D16@185.189.151.222>;tag=260b305c6f7b37c807cb5e3927b9582a
To: <sip:BOBO_D16@185.189.151.222>;tag=z9hG4bK4520303178ac55aaa7db5c740b3f635c
CSeq: 1846984945 SUBSCRIBE
WWW-Authenticate: Digest realm="asterisk",nonce="1618146259/598f751c4b1e1ed0b9def3d8193a0902",opaque="2072424f7a67ee60",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.3.0
Content-Length:  0


<--- Received SIP request (920 bytes) from UDP:129.0.205.28:49101 --->
SUBSCRIBE sip:BOBO_D16@185.189.151.222:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.115:6060;branch=z9hG4bK99bc13d6348c738061615261095fa534;rport
From: <sip:BOBO_D16@185.189.151.222:6060>;tag=260b305c6f7b37c807cb5e3927b9582a
To: <sip:BOBO_D16@185.189.151.222:6060>
Call-ID: 39f9e932d93990fcb552f50ea302923f@192.168.101.115
CSeq: 1846984946 SUBSCRIBE
Contact: <sip:BOBO_D16@192.168.101.115:6060>
Authorization: Digest username="BOBO_D16", realm="asterisk", nonce="1618146259/598f751c4b1e1ed0b9def3d8193a0902", uri="sip:BOBO_D16@185.189.151.222:6060", response="94ae11f700893aa8851434fbb446af56", algorithm=md5, cnonce="4d0f7a9316e045ac04a9e533dcc6c7b3", opaque="2072424f7a67ee60", qop=auth, nc=00000001
Event: message-summary
Supported: eventlist
Expires: 16
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
Max-Forwards: 70
Accept: application/simple-message-summary
Content-Length: 0


<--- Transmitting SIP response (436 bytes) to UDP:129.0.205.28:49101 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.101.115:6060;rport=49101;received=129.0.205.28;branch=z9hG4bK99bc13d6348c738061615261095fa534
Call-ID: 39f9e932d93990fcb552f50ea302923f@192.168.101.115
From: <sip:BOBO_D16@185.189.151.222>;tag=260b305c6f7b37c807cb5e3927b9582a
To: <sip:BOBO_D16@185.189.151.222>;tag=z9hG4bK99bc13d6348c738061615261095fa534
CSeq: 1846984946 SUBSCRIBE
Server: Asterisk PBX 18.3.0
Content-Length:  0


<--- Received SIP response (504 bytes) from UDP:129.0.205.28:49101 --->
SIP/2.0 480 Temporarily not available
Via: SIP/2.0/UDP 185.189.151.222:6060;rport=6060;branch=z9hG4bKPj7f01f1da-016c-44c8-80e6-476254c6ec67
From: <sip:102@185.189.151.222>;tag=7735c908-241a-4b16-b172-e8b80fdf9df2
To: <sip:237659856358@129.0.205.28>;tag=3cc7f8b4da77fff71589681b924af888
Call-ID: 3575c280-e110-4ca9-8534-6cb6a9226bde
CSeq: 7049 INVITE
Contact: <sip:237659856358@192.168.101.115:6060>
Supported: replaces
Reason: GSM;cause=8;text="Operator determined barring"
Content-Length: 0


<--- Transmitting SIP request (431 bytes) to UDP:129.0.205.28:49101 --->
ACK sip:237659856358@129.0.205.28:49101 SIP/2.0
Via: SIP/2.0/UDP 185.189.151.222:6060;rport;branch=z9hG4bKPj7f01f1da-016c-44c8-80e6-476254c6ec67
From: <sip:102@185.189.151.222>;tag=7735c908-241a-4b16-b172-e8b80fdf9df2
To: <sip:237659856358@129.0.205.28>;tag=3cc7f8b4da77fff71589681b924af888
Call-ID: 3575c280-e110-4ca9-8534-6cb6a9226bde
CSeq: 7049 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Length:  0


    -- No one is available to answer at this time (1:0/0/0)
    -- Executing [237659856358@testing:3] Dial("PJSIP/102-00000000", "SIP/BOBO_D16/237659856358") in new stack
[Apr 11 14:04:21] WARNING[2097][C-00000001]: app_dial.c:2598 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
    -- No devices or endpoints to dial (technology/resource)
    -- Auto fallthrough, channel 'PJSIP/102-00000000' status is 'CHANUNAVAIL'
<--- Transmitting SIP response (402 bytes) to UDP:154.72.150.215:42675 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.153.1.160:10197;rport=42675;received=154.72.150.215;branch=z9hG4bK-524287-1---afc2fb4dd695a45e
Call-ID: 3XcPn6ApEkMfoXwz1aqwIg..
From: <sip:102@185.189.151.222>;tag=04346b00
To: <sip:237659856358@185.189.151.222>;tag=6a3d873f-1f52-4188-b052-b1c0734bbe65
CSeq: 2 INVITE
Server: Asterisk PBX 18.3.0
Reason: Q.850;cause=34
Content-Length:  0


<--- Received SIP request (346 bytes) from UDP:154.72.150.215:42675 --->
ACK sip:237659856358@185.189.151.222 SIP/2.0
Via: SIP/2.0/UDP 10.153.1.160:10197;branch=z9hG4bK-524287-1---afc2fb4dd695a45e;rport
Max-Forwards: 70
To: <sip:237659856358@185.189.151.222>;tag=6a3d873f-1f52-4188-b052-b1c0734bbe65
From: <sip:102@185.189.151.222>;tag=04346b00
Call-ID: 3XcPn6ApEkMfoXwz1aqwIg..
CSeq: 2 ACK
Content-Length: 0


<--- Received SIP request (1195 bytes) from UDP:154.72.150.215:42675 --->
REGISTER sip:185.189.151.222 SIP/2.0
Via: SIP/2.0/UDP 10.153.1.160:10197;branch=z9hG4bK-524287-1---baad193705c71767;rport
Max-Forwards: 70
Contact: <sip:102@154.72.150.215:42675>;+sip.instance="<urn:uuid:e20f347c-4e9e-46a3-a10d-272460158a9b>"
To: <sip:102@185.189.151.222>
From: <sip:102@185.189.151.222>;tag=bbb1f92f
Call-ID: x8EV5DgZ5A0K72XehWpGuw..
CSeq: 62 REGISTER
Expires: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH
Supported: replaces, answermode, eventlist, outbound, path
User-Agent: PortSIP UC Client
Authorization: Digest username="102",realm="asterisk",nonce="1618146179/78ed4960bac6c45c518c05a1e14af294",uri="sip:185.189.151.222",response="a364fe3ec95710d23fdf8baaeea8435c",cnonce="8f4acfcc9fd4132159fc2775101b32cb",nc=00000002,qop=auth,algorithm=md5,opaque="0723f7631813b02b"
Allow-Events: hold, talk, conference, dialog
x-p-push: device-os=android;device-uid=frZbXSEjnyI:APA91bGWawhfI7kllH2KBxxQY2YoL87r1KTTyksnpm5FLFvq_YSKRUzaqUR3R1xXR91i060OjVL_eyB4tCak2TTUeYIhVJMBC_xaLWSVi95BYf6aiels57fal1hNbrv_20UomFfIFStu;allow-call-push=true;allow-message-push=true;app-id=com.portsip.portgo
Content-Length: 0


<--- Transmitting SIP response (521 bytes) to UDP:154.72.150.215:42675 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.153.1.160:10197;rport=42675;received=154.72.150.215;branch=z9hG4bK-524287-1---baad193705c71767
Call-ID: x8EV5DgZ5A0K72XehWpGuw..
From: <sip:102@185.189.151.222>;tag=bbb1f92f
To: <sip:102@185.189.151.222>;tag=z9hG4bK-524287-1---baad193705c71767
CSeq: 62 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1618146262/5495feed36adb8eab467bba2ad482f4d",opaque="7d958d2548f1bb3f",stale=true,algorithm=md5,qop="auth"
Server: Asterisk PBX 18.3.0
Content-Length:  0


<--- Received SIP request (1195 bytes) from UDP:154.72.150.215:42675 --->
REGISTER sip:185.189.151.222 SIP/2.0
Via: SIP/2.0/UDP 10.153.1.160:10197;branch=z9hG4bK-524287-1---dab2ee0b2003e463;rport
Max-Forwards: 70
Contact: <sip:102@154.72.150.215:42675>;+sip.instance="<urn:uuid:e20f347c-4e9e-46a3-a10d-272460158a9b>"
To: <sip:102@185.189.151.222>
From: <sip:102@185.189.151.222>;tag=bbb1f92f
Call-ID: x8EV5DgZ5A0K72XehWpGuw..
CSeq: 63 REGISTER
Expires: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH
Supported: replaces, answermode, eventlist, outbound, path
User-Agent: PortSIP UC Client
Authorization: Digest username="102",realm="asterisk",nonce="1618146262/5495feed36adb8eab467bba2ad482f4d",uri="sip:185.189.151.222",response="79e7c5cfdeac18abab54554d3a3c081a",cnonce="d832b4c65c802316a82bd54e8a7138de",nc=00000001,qop=auth,algorithm=md5,opaque="7d958d2548f1bb3f"
Allow-Events: hold, talk, conference, dialog
x-p-push: device-os=android;device-uid=frZbXSEjnyI:APA91bGWawhfI7kllH2KBxxQY2YoL87r1KTTyksnpm5FLFvq_YSKRUzaqUR3R1xXR91i060OjVL_eyB4tCak2TTUeYIhVJMBC_xaLWSVi95BYf6aiels57fal1hNbrv_20UomFfIFStu;allow-call-push=true;allow-message-push=true;app-id=com.portsip.portgo
Content-Length: 0


<--- Transmitting SIP request (641 bytes) to UDP:154.72.150.215:42675 --->
NOTIFY sip:102@154.72.150.215:42675 SIP/2.0
Via: SIP/2.0/UDP 185.189.151.222:6060;rport;branch=z9hG4bKPj5079f09a-100d-4c2a-a488-50d3f872a18a
From: <sip:102@185.189.151.222>;tag=2eae2f45-f33a-4f72-af21-4e406efe553c
To: <sip:102@154.72.150.215>
Contact: <sip:102@185.189.151.222:6060>
Call-ID: 98cce559-7cd3-4914-8693-891d2c6e1942
CSeq: 14218 NOTIFY
Subscription-State: terminated
Event: message-summary
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Type: application/simple-message-summary
Content-Length:    48

Messages-Waiting: no
Voice-Message: 0/0 (0/0)

<--- Transmitting SIP response (454 bytes) to UDP:154.72.150.215:42675 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.153.1.160:10197;rport=42675;received=154.72.150.215;branch=z9hG4bK-524287-1---dab2ee0b2003e463
Call-ID: x8EV5DgZ5A0K72XehWpGuw..
From: <sip:102@185.189.151.222>;tag=bbb1f92f
To: <sip:102@185.189.151.222>;tag=z9hG4bK-524287-1---dab2ee0b2003e463
CSeq: 63 REGISTER
Date: Sun, 11 Apr 2021 13:04:22 GMT
Contact: <sip:102@10.153.1.160:10197>;expires=89
Expires: 90
Server: Asterisk PBX 18.3.0
Content-Length:  0


<--- Received SIP response (390 bytes) from UDP:154.72.150.215:42675 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 185.189.151.222:6060;rport=6060;branch=z9hG4bKPj5079f09a-100d-4c2a-a488-50d3f872a18a
Contact: <sip:10.153.1.160:10197>
To: <sip:102@154.72.150.215>;tag=8efe9705
From: <sip:102@185.189.151.222>;tag=2eae2f45-f33a-4f72-af21-4e406efe553c
Call-ID: 98cce559-7cd3-4914-8693-891d2c6e1942
CSeq: 14218 NOTIFY
User-Agent: PortSIP UC Client
Content-Length: 0

This isn’t a no-way audio case, it is a call that was refused by the mobile operator and never reached the connected state.

I’m a bit confused by the zero content length on outbound INVITE, as I didn’t think Asterisk could be forced to use late offer SDP, and if it could, I wouldn’t have thought it was the default.

My guess is that the mobile operator expects the number to be as you would dial it on a phone, presumably, in this case, starting with +23, rather than just 23.

Thank’s for response
Using 23 or +23 is not an issue
But about

It’s not a problem because operator send ivr to tell the ussue on the route , but there i can’t hear anything

That would have been sent as early media. However, if you don’t send SDP in the INVITE, early media is impossible, and if you had, you would have needed to call Progress() in the dialplan, assuming that the caller could accept early media.

It is possible that something is going wrong whilst generating the SDP, but I’d expect log entries for that. There have been reports that doing allow all can result in strange things happening; you should always restrict the choice of codecs.

I’m not familiar with the ARA schema for PJSIP, so I don’t know how to convert your database select outputs to the minimal equivalent pjsip.conf entries.

Please share me minimum pjsip.conf that you are speaking about and i’ll try it and send you back the result there

https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples

By minimal, I mean that it only contains settings that aren’t defaults.

(Some ARA users use the CLI command to print the configuration, and that contains just too much noise from default values.)

This is the solution
I restrict on ulaw and it work fine now

Really thank you for your help
Not need to allow all codecs on database

It was a very bad idea