Can't make internal calls

Please help!!

I cant make internal call from 917804081 to 917804084

This is my extension.conf
[default]
; Any number of 9 digit begin by 9X
exten = _9[1-8]XXXXXXX,1,verbose( calling to ${EXTEN})
same = n,Dial(PJSIP/${EXTEN},25)

This is my pjsip.conf
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

[917804081]
type = endpoint
transport = transport-udp
context = default
disallow = all
allow = ulaw
aors = 917804081
auth = auth917804081
identify_by = header,username
callerid=917804081

[917804081]
type = aor
max_contacts = 1

[auth917804081]
type=auth
auth_type=userpass
password=917804081
username=917804081

[917804084]
type = endpoint
transport = transport-udp
context = default
disallow = all
allow = ulaw
aors = 917804084
auth = auth917804084
identify_by = header,username
callerid=917804081

[917804084]
type = aor
max_contacts = 1

[auth917804084]
type=auth
auth_type=userpass
password=917804084
username=917804084

These are pjsip logger messages

*CLI>
<— Received SIP request (900 bytes) from UDP:10.10.10.1:5070 —>
INVITE sip:917804084@10.10.10.2 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.1:5070;branch=z9hG4bK1178762566-237
Route: sip:10.10.10.2:5060;lr
Max-Forwards: 70
Accept: application/sdp
Accept: application/dtmf-relay
Accept-Encoding: identity
User-Agent: MitraStar GPT-2541GNAC ES_s00.00_g001_100VNJ0b38_2
Supported: timer,100rel,replaces,message-summary
From: 917804081sip:917804081@10.10.10.2;tag=MSTCUA_1178762905-237
To: sip:917804084@10.10.10.2
Call-ID: 1178742920-237
CSeq: 1 INVITE
Min-SE: 180
Session-Expires: 600;refresher=uac
Contact: 917804081sip:917804081@10.10.10.1:5070
Content-Type: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,OPTIONS,REFER,INFO,PRACK,NOTIFY,MESSAGE
Content-Length: 174

v=0
o=ICF 12345 787 IN IP4 10.10.10.1
s=Session
c=IN IP4 10.10.10.1
t=0 0
m=audio 50000 RTP/AVP 8 0
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20

<— Transmitting SIP response (489 bytes) to UDP:10.10.10.1:5070 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.10.10.1:5070;rport=5070;received=10.10.10.1;branch=z9hG4bK1178762566-237
Call-ID: 1178742920-237
From: “917804081” sip:917804081@10.10.10.2;tag=MSTCUA_1178762905-237
To: sip:917804084@10.10.10.2;tag=z9hG4bK1178762566-237
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1527522011/8c2401d38bdc7d00e8fa33fdc98279b9”,opaque=“40d58f655892acaa”,algorithm=md5,qop=“auth”
Server: Asterisk PBX 15.4.0
Content-Length: 0

<— Received SIP request (389 bytes) from UDP:10.10.10.1:5070 —>
ACK sip:917804084@10.10.10.2 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.1:5070;branch=z9hG4bK1178762566-237
Route: sip:10.10.10.2:5060;lr
Max-Forwards: 70
From: 917804081sip:917804081@10.10.10.2;tag=MSTCUA_1178762905-237
To: sip:917804084@10.10.10.2;tag=z9hG4bK1178762566-237
Call-ID: 1178742920-237
CSeq: 1 ACK
Contact: 917804081sip:917804081@10.10.10.1:5070
Content-Length: 0

<— Received SIP request (1169 bytes) from UDP:10.10.10.1:5070 —>
INVITE sip:917804084@10.10.10.2 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.1:5070;branch=z9hG4bK1178789347-237
Route: sip:10.10.10.2:5060;lr
Max-Forwards: 70
Accept: application/sdp
Accept: application/dtmf-relay
Accept-Encoding: identity
User-Agent: MitraStar GPT-2541GNAC ES_s00.00_g001_100VNJ0b38_2
Supported: timer,100rel,replaces,message-summary
From: 917804081sip:917804081@10.10.10.2;tag=MSTCUA_1178762905-237
To: sip:917804084@10.10.10.2
Call-ID: 1178742920-237
CSeq: 2 INVITE
Min-SE: 180
Session-Expires: 600;refresher=uac
Contact: 917804081sip:917804081@10.10.10.1:5070
Content-Type: application/sdp
Authorization: Digest username=“917804081”,realm=“asterisk”,nonce=“1527522011/8c2401d38bdc7d00e8fa33fdc98279b9”,uri=“sip:917804084@10.10.10.2”,response=“7fa5167f1bd66144581307958129002d”,algorithm=md5,cnonce=“1178787496”,opaque=“40d58f655892acaa”,qop=auth,nc=00000001
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,OPTIONS,REFER,INFO,PRACK,NOTIFY,MESSAGE
Content-Length: 174

v=0
o=ICF 12345 787 IN IP4 10.10.10.1
s=Session
c=IN IP4 10.10.10.1
t=0 0
m=audio 50000 RTP/AVP 8 0
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20

== Setting global variable ‘SIPDOMAIN’ to ‘10.10.10.2’
<— Transmitting SIP response (310 bytes) to UDP:10.10.10.1:5070 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.1:5070;rport=5070;received=10.10.10.1;branch=z9hG4bK1178789347-237
Call-ID: 1178742920-237
From: “917804081” sip:917804081@10.10.10.2;tag=MSTCUA_1178762905-237
To: sip:917804084@10.10.10.2
CSeq: 2 INVITE
Server: Asterisk PBX 15.4.0
Content-Length: 0

-- Executing [917804084@default:1] Verbose("PJSIP/917804081-00000003", " calling to 917804084") in new stack

calling to 917804084
– Executing [917804084@default:2] Dial(“PJSIP/917804081-00000003”, “PJSIP/917804084,25”) in new stack
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘PJSIP/917804081-00000003’ status is ‘CHANUNAVAIL’
<— Transmitting SIP response (383 bytes) to UDP:10.10.10.1:5070 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.10.10.1:5070;rport=5070;received=10.10.10.1;branch=z9hG4bK1178789347-237
Call-ID: 1178742920-237
From: “917804081” sip:917804081@10.10.10.2;tag=MSTCUA_1178762905-237
To: sip:917804084@10.10.10.2;tag=iZBnLppQ.yEp4DXxuG45KRs7NP-7YHwi
CSeq: 2 INVITE
Server: Asterisk PBX 15.4.0
Reason: Q.850;cause=3
Content-Length: 0

<— Received SIP request (669 bytes) from UDP:10.10.10.1:5070 —>
ACK sip:917804084@10.10.10.2 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.1:5070;branch=z9hG4bK1178789347-237
Route: sip:10.10.10.2:5060;lr
Max-Forwards: 70
Authorization: Digest username=“917804081”,realm=“asterisk”,nonce=“1527522011/8c2401d38bdc7d00e8fa33fdc98279b9”,uri=“sip:917804084@10.10.10.2”,response=“7fa5167f1bd66144581307958129002d”,algorithm=md5,cnonce=“1178787496”,opaque=“40d58f655892acaa”,qop=auth,nc=00000001
From: 917804081sip:917804081@10.10.10.2;tag=MSTCUA_1178762905-237
To: sip:917804084@10.10.10.2;tag=iZBnLppQ.yEp4DXxuG45KRs7NP-7YHwi
Call-ID: 1178742920-237
CSeq: 2 ACK
Contact: 917804081sip:917804081@10.10.10.1:5070
Content-Length: 0

what is the reason??
thanks in advance.

Accrording to the below excerpt from your configuration, 917804084 is the dynamic endpoint which should be registered on Asterisk before it can be called:

[917804084]
type = aor
max_contacts = 1

However, it looks like it is not available:

== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘PJSIP/917804081-00000003’ status is ‘CHANUNAVAIL’

Are you sure that 917804084 is registered?

Please provide
pjsip show contacts

Yes, both are registered.
see this logs to force re-register on both ip-phones:

*CLI> pjsip set logger on
PJSIP Logging enabled
*CLI> <— Received SIP request (710 bytes) from UDP:10.10.10.1:5070 —>
REGISTER sip:10.10.10.2 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.1:5070;branch=z9hG4bK3586642593-237
Route: sip:10.10.10.2:5060;lr
Max-Forwards: 70
Accept-Encoding: identity
Accept: application/simple-message-summary
Accept: application/sdp
Accept: application/dtmf-relay
Accept: application/dtmf
User-Agent: MitraStar GPT-2541GNAC ES_s00.00_g001_100VNJ0b38_2
Supported: replaces,path
From: 917804081sip:917804081@10.10.10.2;tag=MSTCUA_3586642860-237
To: 917804081sip:917804081@10.10.10.2
Call-ID: 3586624181-237
CSeq: 1 REGISTER
Expires: 600
Contact: sip:917804081@10.10.10.1:5070;expires=600
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
Content-Length: 0

<— Transmitting SIP response (503 bytes) to UDP:10.10.10.1:5070 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.10.10.1:5070;rport=5070;received=10.10.10.1;branch=z9hG4bK3586642593-237
Call-ID: 3586624181-237
From: “917804081” sip:917804081@10.10.10.2;tag=MSTCUA_3586642860-237
To: “917804081” sip:917804081@10.10.10.2;tag=z9hG4bK3586642593-237
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1527601720/7128e23ac2a34856f053599f60403fbe”,opaque=“328dbd967d5a19f2”,algorithm=md5,qop=“auth”
Server: Asterisk PBX 15.4.0
Content-Length: 0

<— Received SIP request (969 bytes) from UDP:10.10.10.1:5070 —>
REGISTER sip:10.10.10.2 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.1:5070;branch=z9hG4bK3586660582-237
Route: sip:10.10.10.2:5060;lr
Max-Forwards: 70
Accept-Encoding: identity
Accept: application/simple-message-summary
Accept: application/sdp
Accept: application/dtmf-relay
Accept: application/dtmf
User-Agent: MitraStar GPT-2541GNAC ES_s00.00_g001_100VNJ0b38_2
Supported: replaces,path
From: 917804081sip:917804081@10.10.10.2;tag=MSTCUA_3586642860-237
To: 917804081sip:917804081@10.10.10.2
Call-ID: 3586624181-237
CSeq: 2 REGISTER
Authorization: Digest username=“917804081”,realm=“asterisk”,nonce=“1527601720/7128e23ac2a34856f053599f60403fbe”,uri=“sip:10.10.10.2”,response=“e0c2c1f39d2d187e367ce2ead23b8ad9”,algorithm=md5,cnonce=“3586659011”,opaque=“328dbd967d5a19f2”,qop=auth,nc=00000001
Expires: 600
Contact: sip:917804081@10.10.10.1:5070;expires=600
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
Content-Length: 0

<— Transmitting SIP response (397 bytes) to UDP:10.10.10.1:5070 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.1:5070;rport=5070;received=10.10.10.1;branch=z9hG4bK3586660582-237
Call-ID: 3586624181-237
From: “917804081” sip:917804081@10.10.10.2;tag=MSTCUA_3586642860-237
To: “917804081” sip:917804081@10.10.10.2;tag=z9hG4bK3586660582-237
CSeq: 2 REGISTER
Date: Tue, 29 May 2018 13:48:40 GMT
Expires: 600
Server: Asterisk PBX 15.4.0
Content-Length: 0

<— Received SIP request (677 bytes) from UDP:10.10.10.4:5070 —>
REGISTER sip:10.10.10.2 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.4:5070;branch=z9hG4bK2924517438-237
Max-Forwards: 70
Accept-Encoding: identity
Accept: application/simple-message-summary
Accept: application/sdp
Accept: application/dtmf-relay
Accept: application/dtmf
User-Agent: MitraStar GPT-2541GNAC ES_s00.00_g001_100VNJ0b38_2
Supported: replaces,path
From: 917804084sip:917804084@10.10.10.2;tag=MSTCUA_2924517896-237
To: 917804084sip:917804084@10.10.10.2
Call-ID: 2924499589-237
CSeq: 1 REGISTER
Expires: 600
Contact: sip:917804084@10.10.10.4:5070;expires=600
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
Content-Length: 0

<— Transmitting SIP response (503 bytes) to UDP:10.10.10.4:5070 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.10.10.4:5070;rport=5070;received=10.10.10.4;branch=z9hG4bK2924517438-237
Call-ID: 2924499589-237
From: “917804084” sip:917804084@10.10.10.2;tag=MSTCUA_2924517896-237
To: “917804084” sip:917804084@10.10.10.2;tag=z9hG4bK2924517438-237
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1527601726/ba5fbe5d9dd78608e6838b1d58e70611”,opaque=“5e583c532fbf63d2”,algorithm=md5,qop=“auth”
Server: Asterisk PBX 15.4.0
Content-Length: 0

<— Received SIP request (969 bytes) from UDP:10.10.10.4:5070 —>
REGISTER sip:10.10.10.2 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.4:5070;branch=z9hG4bK2924534066-237
Route: sip:10.10.10.2:5060;lr
Max-Forwards: 70
Accept-Encoding: identity
Accept: application/simple-message-summary
Accept: application/sdp
Accept: application/dtmf-relay
Accept: application/dtmf
User-Agent: MitraStar GPT-2541GNAC ES_s00.00_g001_100VNJ0b38_2
Supported: replaces,path
From: 917804084sip:917804084@10.10.10.2;tag=MSTCUA_2924517896-237
To: 917804084sip:917804084@10.10.10.2
Call-ID: 2924499589-237
CSeq: 2 REGISTER
Authorization: Digest username=“917804084”,realm=“asterisk”,nonce=“1527601726/ba5fbe5d9dd78608e6838b1d58e70611”,uri=“sip:10.10.10.2”,response=“fa76d6a0d797ebe46cfd4c7c9f6e5411”,algorithm=md5,cnonce=“2924532387”,opaque=“5e583c532fbf63d2”,qop=auth,nc=00000001
Expires: 600
Contact: sip:917804084@10.10.10.4:5070;expires=600
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
Content-Length: 0

<— Transmitting SIP response (397 bytes) to UDP:10.10.10.4:5070 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.4:5070;rport=5070;received=10.10.10.4;branch=z9hG4bK2924534066-237
Call-ID: 2924499589-237
From: “917804084” sip:917804084@10.10.10.2;tag=MSTCUA_2924517896-237
To: “917804084” sip:917804084@10.10.10.2;tag=z9hG4bK2924534066-237
CSeq: 2 REGISTER
Date: Tue, 29 May 2018 13:48:46 GMT
Expires: 600
Server: Asterisk PBX 15.4.0
Content-Length: 0

*CLI> pjsip show contacts
No objects found.

*CLI> pjsip show contact 917804081
Unable to find object 917804081.

*CLI> pjsip show contact 917804084
Unable to find object 917804084.

How are you storing things? Have you modified sorcery.conf and are using realtime or something else?

No sorcery modified. only these files in etc/asterisk

drwxr-xr-x 4 root root 4096 may 28 18:51 .
drwxr-xr-x. 184 root root 12288 may 28 17:59 …
-rw-r–r-- 1 adsl adsl 750 may 28 18:41 extensions.conf
-rw-r–r-- 1 adsl adsl 5384 may 17 18:32 http.conf
-rw-r–r-- 1 adsl adsl 2620 may 17 15:36 modules.conf
-rw-r–r-- 1 adsl adsl 2819 may 28 18:51 pjsip.conf

Are you selectively loading modules or something… it’s entirely possible that you’ve slimmed things down so much that it can’t persist data.

Modules.conf contain these:

[modules]
autoload = no

; This is a minimal module load. We are loading only the modules required for
; the Asterisk features used in the Super Awesome Company configuration.

; Applications

load = app_bridgewait.so
load = app_dial.so
load = app_playback.so
load = app_stack.so
load = app_verbose.so
load = app_voicemail.so
load = app_directory.so
load = app_confbridge.so

; Bridging

load = bridge_builtin_features.so
load = bridge_builtin_interval_features.so
load = bridge_holding.so
load = bridge_native_rtp.so
load = bridge_simple.so
load = bridge_softmix.so

; Call Detail Records

load = cdr_custom.so

; Channel Drivers

load = chan_bridge_media.so
load = chan_pjsip.so

; Codecs

load = codec_gsm.so
load = codec_resample.so
load = codec_ulaw.so
load = codec_g722.so

; Formats

load = format_gsm.so
load = format_pcm.so
load = format_wav_gsm.so
load = format_wav.so

; Functions

load = func_callerid.so
load = func_cdr.so
load = func_pjsip_endpoint.so
load = func_sorcery.so
load = func_devstate.so
load = func_strings.so

; Core/PBX

load = pbx_config.so

; Resources

load = res_musiconhold.so
load = res_pjproject.so
load = res_pjsip_acl.so
load = res_pjsip_authenticator_digest.so
load = res_pjsip_caller_id.so
load = res_pjsip_dialog_info_body_generator.so
load = res_pjsip_diversion.so
load = res_pjsip_dtmf_info.so
load = res_pjsip_endpoint_identifier_anonymous.so
load = res_pjsip_endpoint_identifier_ip.so
load = res_pjsip_endpoint_identifier_user.so
load = res_pjsip_exten_state.so
load = res_pjsip_header_funcs.so
load = res_pjsip_logger.so
load = res_pjsip_messaging.so
load = res_pjsip_mwi_body_generator.so
load = res_pjsip_mwi.so
load = res_pjsip_nat.so
load = res_pjsip_notify.so
load = res_pjsip_one_touch_record_info.so
load = res_pjsip_outbound_authenticator_digest.so
load = res_pjsip_outbound_publish.so
load = res_pjsip_outbound_registration.so
load = res_pjsip_path.so
load = res_pjsip_pidf_body_generator.so
load = res_pjsip_pidf_digium_body_supplement.so
load = res_pjsip_pidf_eyebeam_body_supplement.so
load = res_pjsip_publish_asterisk.so
load = res_pjsip_pubsub.so
load = res_pjsip_refer.so
load = res_pjsip_registrar_expire.so
load = res_pjsip_registrar.so
load = res_pjsip_rfc3326.so
load = res_pjsip_sdp_rtp.so
load = res_pjsip_send_to_voicemail.so
load = res_pjsip_session.so
load = res_pjsip.so
load = res_pjsip_t38.so
load = res_pjsip_transport_websocket.so
load = res_pjsip_xpidf_body_generator.so
load = res_rtp_asterisk.so
load = res_sorcery_astdb.so
load = res_sorcery_config.so
load = res_sorcery_memory.so
load = res_sorcery_realtime.so
load = res_timing_timerfd.so

I would suggest autoloading first and seeing if it works. If it does then the problem is in the set of modules that are loaded. Nothing immediately stands out as missing from the list.

my new modules.conf

[modules]
autoload=yes

CLI> module show
Module Description Use Count Status Support Level
app_adsiprog.so Asterisk ADSI Programming Application 0 Running extended
app_agent_pool.so Call center agent pool applications 0 Not Running core
app_alarmreceiver.so Alarm Receiver for Asterisk 0 Not Running extended
app_amd.so Answering Machine Detection Application 0 Not Running extended
app_authenticate.so Authentication Application 0 Running core
app_bridgeaddchan.so Bridge Add Channel Application 0 Running core
app_bridgewait.so Place the channel into a holding bridge 0 Running core
app_cdr.so Tell Asterisk to not maintain a CDR for 0 Running core
app_celgenuserevent.so Generate an User-Defined CEL event 0 Running core
app_chanisavail.so Check channel availability 0 Running extended
app_channelredirect.so Redirects a given channel to a dialplan 0 Running core
app_chanspy.so Listen to the audio of an active channel 0 Running core
app_confbridge.so Conference Bridge Application 0 Not Running core
app_controlplayback.so Control Playback Application 0 Running core
app_db.so Database Access Functions 0 Running core
app_dial.so Dialing Application 0 Running core
app_dictate.so Virtual Dictation Machine 0 Running extended
app_directed_pickup.so Directed Call Pickup Application 0 Running core
app_directory.so Extension Directory 0 Running core
app_disa.so DISA (Direct Inward System Access) Appli 0 Running core
app_dumpchan.so Dump Info About The Calling Channel 0 Running core
app_echo.so Simple Echo Application 0 Running core
app_exec.so Executes dialplan applications 0 Running core
app_externalivr.so External IVR Interface Application 0 Running extended
app_festival.so Simple Festival Interface 0 Not Running extended
app_followme.so Find-Me/Follow-Me Application 0 Not Running core
app_forkcdr.so Fork The CDR into 2 separate entities 0 Running core
app_getcpeid.so Get ADSI CPE ID 0 Running extended
app_ices.so Encode and Stream via icecast and ices 0 Running extended
app_image.so Image Transmission Application 0 Running extended
app_macro.so Extension Macros 0 Running core
app_milliwatt.so Digital Milliwatt (mu-law) Test Applicat 0 Running core
app_minivm.so Mini VoiceMail (A minimal Voicemail e-ma 0 Running extended
app_mixmonitor.so Mixed Audio Monitoring Application 0 Running core
app_morsecode.so Morse code 0 Running extended
app_mp3.so Silly MP3 Application 0 Running extended
app_nbscat.so Silly NBS Stream Application 0 Running extended
app_originate.so Originate call 0 Running core
app_page.so Page Multiple Phones 0 Running core
app_playback.so Sound File Playback Application 0 Running core
app_playtones.so Playtones Application 0 Running core
app_privacy.so Require phone number to be entered, if n 0 Running core
app_queue.so True Call Queueing 0 Not Running core
app_read.so Read Variable Application 0 Running core
app_readexten.so Read and evaluate extension validity 0 Running core
app_record.so Trivial Record Application 0 Running core
app_sayunixtime.so Say time 0 Running core
app_senddtmf.so Send DTMF digits Application 0 Running core
app_sendtext.so Send Text Applications 0 Running core
app_sms.so SMS/PSTN handler 0 Running extended
app_softhangup.so Hangs up the requested channel 0 Running core
app_speech_utils.so Dialplan Speech Applications 0 Running core
app_stack.so Dialplan subroutines (Gosub, Return, etc 0 Running core
app_stasis.so Stasis dialplan application 0 Running core
app_stream_echo.so Stream Echo Application 0 Running core
app_system.so Generic System() application 0 Running core
app_talkdetect.so Playback with Talk Detection 0 Running extended
app_test.so Interface Test Application 0 Running extended
app_transfer.so Transfers a caller to another extension 0 Running core
app_url.so Send URL Applications 0 Running extended
app_userevent.so Custom User Event Application 0 Running core
app_verbose.so Send verbose output 0 Running core
app_voicemail.so Comedian Mail (Voicemail System) 0 Running core
app_waitforring.so Waits until first ring after time 0 Running extended
app_waitforsilence.so Wait For Silence/Noise 0 Running extended
app_waituntil.so Wait until specified time 0 Running core
app_while.so While Loops and Conditional Execution 0 Running core
app_zapateller.so Block Telemarketers with Special Informa 0 Running extended
bridge_builtin_features.so Built in bridging features 1 Running core
bridge_builtin_interval_features.so Built in bridging interval features 0 Running core
bridge_holding.so Holding bridge module 0 Running core
bridge_native_rtp.so Native RTP bridging module 0 Running core
bridge_simple.so Simple two channel bridging module 0 Running core
bridge_softmix.so Multi-party software based channel mixin 0 Running core
cdr_csv.so Comma Separated Values CDR Backend 0 Not Running extended
cdr_custom.so Customizable Comma Separated Values CDR 0 Running core
cdr_manager.so Asterisk Manager Interface CDR Backend 0 Not Running core
cdr_sqlite3_custom.so SQLite3 Custom CDR Module 0 Not Running extended
cdr_syslog.so Customizable syslog CDR Backend 0 Not Running core
cel_custom.so Customizable Comma Separated Values CEL 0 Running core
cel_manager.so Asterisk Manager Interface CEL Backend 0 Not Running core
cel_sqlite3_custom.so SQLite3 Custom CEL Module 0 Not Running extended
chan_alsa.so ALSA Console Channel Driver 0 Not Running extended
chan_bridge_media.so Bridge Media Channel Driver 0 Running core
chan_iax2.so Inter Asterisk eXchange (Ver 2) 0 Not Running core
chan_mgcp.so Media Gateway Control Protocol (MGCP) 0 Running extended
chan_oss.so OSS Console Channel Driver 0 Not Running extended
chan_phone.so Linux Telephony API Support 0 Not Running extended
chan_pjsip.so PJSIP Channel Driver 0 Running core
chan_rtp.so RTP Media Channel 0 Running core
chan_sip.so Session Initiation Protocol (SIP) 0 Not Running core
chan_skinny.so Skinny Client Control Protocol (Skinny) 0 Not Running extended
chan_unistim.so UNISTIM Protocol (USTM) 0 Not Running extended
codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 Running core
codec_adpcm.so Adaptive Differential PCM Coder/Decoder 0 Running core
codec_alaw.so A-law Coder/Decoder 0 Running core
codec_g722.so ITU G.722-64kbps G722 Transcoder 0 Running core
codec_g726.so ITU G.726-32kbps G726 Transcoder 0 Running core
codec_gsm.so GSM Coder/Decoder 0 Running core
codec_ilbc.so iLBC Coder/Decoder 0 Running core
codec_lpc10.so LPC10 2.4kbps Coder/Decoder 0 Running core
codec_resample.so SLIN Resampling Codec 0 Running core
codec_ulaw.so mu-Law Coder/Decoder 0 Running core
format_g719.so ITU G.719 0 Running core
format_g723.so G.723.1 Simple Timestamp File Format 0 Running core
format_g726.so Raw G.726 (16/24/32/40kbps) data 0 Running core
format_g729.so Raw G.729 data 0 Running core
format_gsm.so Raw GSM data 0 Running core
format_h263.so Raw H.263 data 0 Running core
format_h264.so Raw H.264 data 0 Running core
format_ilbc.so Raw iLBC data 0 Running core
format_jpeg.so jpeg (joint picture experts group) image 0 Running extended
format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0 Running core
format_siren14.so ITU G.722.1 Annex C (Siren14, licensed f 0 Running core
format_siren7.so ITU G.722.1 (Siren7, licensed from Polyc 0 Running core
format_sln.so Raw Signed Linear Audio support (SLN) 8k 0 Running core
format_vox.so Dialogic VOX (ADPCM) File Format 0 Running extended
format_wav.so Microsoft WAV/WAV16 format (8kHz/16kHz S 0 Running core
format_wav_gsm.so Microsoft WAV format (Proprietary GSM) 0 Running core
func_aes.so AES dialplan functions 0 Running core
func_base64.so base64 encode/decode dialplan functions 0 Running core
func_blacklist.so Look up Caller
ID name/number from black 0 Running core
func_callcompletion.so Call Control Configuration Function 0 Running core
func_callerid.so Party ID related dialplan functions (Cal 0 Running core
func_cdr.so Call Detail Record (CDR) dialplan functi 0 Running core
func_channel.so Channel information dialplan functions 0 Running core
func_config.so Asterisk configuration file variable acc 0 Running core
func_curl.so Load external URL 0 Running core
func_cut.so Cut out information from a string 0 Running core
func_db.so Database (astdb) related dialplan functi 0 Running core
func_devstate.so Gets or sets a device state in the dialp 0 Running core
func_dialgroup.so Dialgroup dialplan function 0 Running core
func_dialplan.so Dialplan Context/Extension/Priority Chec 0 Running core
func_enum.so ENUM related dialplan functions 0 Running core
func_env.so Environment/filesystem dialplan function 0 Running core
func_extstate.so Gets an extension’s state in the dialpla 0 Running core
func_frame_trace.so Frame Trace for internal ast_frame debug 0 Running extended
func_global.so Variable dialplan functions 0 Running core
func_groupcount.so Channel group dialplan functions 0 Running core
func_hangupcause.so HANGUPCAUSE related functions and applic 0 Running core
func_holdintercept.so Hold interception dialplan function 0 Running core
func_iconv.so Charset conversions 0 Running core
func_jitterbuffer.so Jitter buffer for read side of channel. 0 Running core
func_lock.so Dialplan mutexes 0 Running core
func_logic.so Logical dialplan functions 0 Running core
func_math.so Mathematical dialplan function 0 Running core
func_md5.so MD5 digest dialplan functions 0 Running core
func_module.so Checks if Asterisk module is loaded in m 0 Running core
func_periodic_hook.so Periodic dialplan hooks. 1 Running core
func_pitchshift.so Audio Effects Dialplan Functions 0 Running extended
func_pjsip_aor.so Get information about a PJSIP AOR 0 Running core
func_pjsip_contact.so Get information about a PJSIP contact 0 Running core
func_pjsip_endpoint.so Get information about a PJSIP endpoint 0 Running core
func_presencestate.so Gets or sets a presence state in the dia 0 Running core
func_rand.so Random number dialplan function 0 Running core
func_realtime.so Read/Write/Store/Destroy values from a R 0 Running core
func_sha1.so SHA-1 computation dialplan function 0 Running core
func_shell.so Collects the output generated by a comma 0 Running core
func_sorcery.so Get a field from a sorcery object 0 Running core
func_sprintf.so SPRINTF dialplan function 0 Running core
func_srv.so SRV related dialplan functions 0 Running core
func_strings.so String handling dialplan functions 0 Running core
func_sysinfo.so System information related functions 0 Running core
func_talkdetect.so Talk detection dialplan function 0 Running core
func_timeout.so Channel timeout dialplan functions 0 Running core
func_uri.so URI encode/decode dialplan functions 0 Running core
func_version.so Get Asterisk Version/Build Info 0 Running core
func_vmcount.so Indicator for whether a voice mailbox ha 0 Running core
func_volume.so Technology independent volume control 0 Running core
pbx_ael.so Asterisk Extension Language Compiler 0 Not Running extended
pbx_config.so Text Extension Configuration 0 Running core
pbx_dundi.so Distributed Universal Number Discovery ( 0 Not Running extended
pbx_loopback.so Loopback Switch 0 Running core
pbx_realtime.so Realtime Switch 0 Running extended
pbx_spool.so Outgoing Spool Support 0 Not Running core
res_adsi.so ADSI Resource 0 Running core
res_ael_share.so share-able code for AEL 0 Running extended
res_agi.so Asterisk Gateway Interface (AGI) 2 Running core
res_ari.so Asterisk RESTful Interface 0 Not Running core
res_ari_applications.so RESTful API module - Stasis application 0 Not Running core
res_ari_asterisk.so RESTful API module - Asterisk resources 0 Not Running core
res_ari_bridges.so RESTful API module - Bridge resources 0 Not Running core
res_ari_channels.so RESTful API module - Channel resources 0 Not Running core
res_ari_device_states.so RESTful API module - Device state resour 0 Not Running core
res_ari_endpoints.so RESTful API module - Endpoint resources 0 Not Running core
res_ari_events.so RESTful API module - WebSocket resource 0 Not Running core
res_ari_model.so ARI Model validators 0 Running core
res_ari_playbacks.so RESTful API module - Playback control re 0 Not Running core
res_ari_recordings.so RESTful API module - Recording resources 0 Not Running core
res_ari_sounds.so RESTful API module - Sound resources 0 Not Running core
res_calendar.so Asterisk Calendar integration 0 Not Running extended
res_clialiases.so CLI Aliases 0 Running core
res_clioriginate.so Call origination and redirection from th 0 Running core
res_config_curl.so Realtime Curl configuration 0 Running core
res_config_ldap.so LDAP realtime interface 0 Running extended
res_config_sqlite.so Realtime SQLite configuration 0 Not Running extended
res_config_sqlite3.so SQLite 3 realtime config engine 0 Running core
res_convert.so File format conversion CLI command 0 Running core
res_crypto.so Cryptographic Digital Signatures 1 Running core
res_curl.so cURL Resource Module 0 Running core
res_fax.so Generic FAX Applications 0 Running core
res_format_attr_celt.so CELT Format Attribute Module 1 Running core
res_format_attr_g729.so G.729 Format Attribute Module 1 Running core
res_format_attr_h263.so H.263 Format Attribute Module 1 Running core
res_format_attr_h264.so H.264 Format Attribute Module 1 Running core
res_format_attr_ilbc.so iLBC Format Attribute Module 1 Running core
res_format_attr_opus.so Opus Format Attribute Module 1 Running core
res_format_attr_silk.so SILK Format Attribute Module 1 Running core
res_format_attr_siren14.so Siren14 Format Attribute Module 1 Running core
res_format_attr_siren7.so Siren7 Format Attribute Module 1 Running core
res_format_attr_vp8.so VP8 Format Attribute Module 1 Running core
res_hep.so HEPv3 API 0 Not Running extended
res_hep_pjsip.so PJSIP HEPv3 Logger 0 Not Running extended
res_hep_rtcp.so RTCP HEPv3 Logger 0 Not Running extended
res_http_media_cache.so HTTP Media Cache Backend 1 Running core
res_http_websocket.so HTTP WebSocket Support 2 Running extended
res_limit.so Resource limits 0 Running core
res_manager_devicestate.so Manager Device State Topic Forwarder 0 Running core
res_manager_presencestate.so Manager Presence State Topic Forwarder 0 Running core
res_monitor.so Call Monitoring Resource 1 Running core
res_musiconhold.so Music On Hold Resource 0 Running core
res_mutestream.so Mute audio stream resources 0 Running core
res_parking.so Call Parking Resource 0 Not Running core
res_phoneprov.so HTTP Phone Provisioning 0 Not Running extended
res_pjproject.so PJPROJECT Log and Utility Support 1 Running core
res_pjsip.so Basic SIP resource 44 Running core
res_pjsip_acl.so PJSIP ACL Resource 0 Running core
res_pjsip_authenticator_digest.so PJSIP authentication resource 0 Running core
res_pjsip_caller_id.so PJSIP Caller ID Support 0 Running core
res_pjsip_config_wizard.so PJSIP Config Wizard 1 Running core
res_pjsip_dialog_info_body_generator.so PJSIP Extension State Dialog Info+XML Pr 0 Running core
res_pjsip_diversion.so PJSIP Add Diversion Header Support 0 Running core
res_pjsip_dlg_options.so SIP OPTIONS in dialog handler 0 Running core
res_pjsip_dtmf_info.so PJSIP DTMF INFO Support 0 Running core
res_pjsip_empty_info.so PJSIP Empty INFO Support 0 Running core
res_pjsip_endpoint_identifier_anonymous.so PJSIP Anonymous endpoint identifier 0 Running core
res_pjsip_endpoint_identifier_ip.so PJSIP IP endpoint identifier 0 Running core
res_pjsip_endpoint_identifier_user.so PJSIP username endpoint identifier 0 Running core
res_pjsip_exten_state.so PJSIP Extension State Notifications 0 Running core
res_pjsip_header_funcs.so PJSIP Header Functions 0 Running core
res_pjsip_history.so PJSIP History 0 Running extended
res_pjsip_logger.so PJSIP Packet Logger 0 Running core
res_pjsip_messaging.so PJSIP Messaging Support 0 Running core
res_pjsip_mwi.so PJSIP MWI resource 0 Running core
res_pjsip_mwi_body_generator.so PJSIP MWI resource 0 Running core
res_pjsip_nat.so PJSIP NAT Support 0 Running core
res_pjsip_notify.so CLI/AMI PJSIP NOTIFY Support 0 Not Running core
res_pjsip_one_touch_record_info.so PJSIP INFO One Touch Recording Support 0 Running core
res_pjsip_outbound_authenticator_digest.so PJSIP authentication resource 0 Running core
res_pjsip_outbound_publish.so PJSIP Outbound Publish Support 4 Running core
res_pjsip_outbound_registration.so PJSIP Outbound Registration Support 0 Running core
res_pjsip_path.so PJSIP Path Header Support 0 Running core
res_pjsip_phoneprov_provider.so PJSIP Phoneprov Provider 0 Not Running extended
res_pjsip_pidf_body_generator.so PJSIP Extension State PIDF Provider 0 Running core
res_pjsip_pidf_digium_body_supplement.so PJSIP PIDF Digium presence supplement 0 Running core
res_pjsip_pidf_eyebeam_body_supplement.so PJSIP PIDF Eyebeam supplement 0 Running core
res_pjsip_publish_asterisk.so PJSIP Asterisk Event PUBLISH Support 0 Running core
res_pjsip_pubsub.so PJSIP event resource 6 Running core
res_pjsip_refer.so PJSIP Blind and Attended Transfer Suppor 1 Running core
res_pjsip_registrar.so PJSIP Registrar Support 0 Running core
res_pjsip_rfc3326.so PJSIP RFC3326 Support 0 Running core
res_pjsip_sdp_rtp.so PJSIP SDP RTP/AVP stream handler 0 Running core
res_pjsip_send_to_voicemail.so PJSIP REFER Send to Voicemail Support 0 Running core
res_pjsip_session.so PJSIP Session resource 4 Running core
res_pjsip_sips_contact.so UAC SIPS Contact support 0 Running core
res_pjsip_t38.so PJSIP T.38 UDPTL Support 0 Running core
res_pjsip_transport_websocket.so PJSIP WebSocket Transport Support 0 Running core
res_pjsip_xpidf_body_generator.so PJSIP Extension State PIDF Provider 0 Running core
res_realtime.so Realtime Data Lookup/Rewrite 0 Running core
res_rtp_asterisk.so Asterisk RTP Stack 0 Running core
res_rtp_multicast.so Multicast RTP Engine 0 Running core
res_security_log.so Security Event Logging 0 Running core
res_smdi.so Simplified Message Desk Interface (SMDI) 0 Not Running extended
res_snmp.so SNMP [Sub]Agent for Asterisk 0 Not Running extended
res_sorcery_astdb.so Sorcery Astdb Object Wizard 2 Running core
res_sorcery_config.so Sorcery Configuration File Object Wizard 16 Running core
res_sorcery_memory.so Sorcery In-Memory Object Wizard 8 Running core
res_sorcery_memory_cache.so Sorcery Memory Cache Object Wizard 0 Running core
res_sorcery_realtime.so Sorcery Realtime Object Wizard 0 Running core
res_speech.so Generic Speech Recognition API 0 Running core
res_stasis.so Stasis application support 2 Running core
res_stasis_answer.so Stasis application answer support 0 Running core
res_stasis_device_state.so Stasis application device state support 0 Running core
res_stasis_playback.so Stasis application playback support 0 Running core
res_stasis_recording.so Stasis application recording support 0 Running core
res_stasis_snoop.so Stasis application snoop support 0 Running core
res_statsd.so Statsd client support 0 Not Running extended
res_stun_monitor.so STUN Network Monitor 0 Not Running core
res_timing_pthread.so pthread Timing Interface 0 Running extended
res_timing_timerfd.so Timerfd Timing Interface 0 Running core
290 modules loaded
*CLI>

*CLI> pjsip show contacts
No objects found.

contacts still are missing.

What does “database show” show? Does Asterisk have permissions to create and use /var/lib/asterisk/astdb.sqlite3?

List of directory:
[root@localhost asterisk]# ls -al
total 68
drwxr-xr-x 14 root root 4096 may 29 17:48 .
drwxr-xr-x. 75 root root 4096 may 14 19:13 …
drwxr-xr-x 2 root root 4096 may 14 19:13 agi-bin
-rw-r–r-- 1 root root 12288 may 29 17:48 astdb.sqlite3
drwxr-xr-x 3 root root 4096 may 14 19:13 documentation
drwxr-xr-x 3 root root 4096 may 14 19:13 firmware
drwxr-xr-x 2 root root 4096 may 14 19:13 images
drwxr-xr-x 2 root root 4096 may 14 19:13 keys
drwxr-xr-x 2 root root 4096 may 14 19:13 moh
drwxr-xr-x 2 root root 4096 may 14 19:13 phoneprov
drwxr-xr-x 2 root root 4096 may 14 19:13 rest-api
drwxr-xr-x 2 root root 4096 may 14 19:13 scripts
drwxr-xr-x 3 root root 4096 may 14 19:13 sounds
drwxr-xr-x 2 root root 4096 may 14 19:13 static-http
drwxr-xr-x 3 root root 4096 may 14 19:13 third-party

asterisk run as root

*CLI> database show
/pbx/UUID : b3793bf5-c2f0-4a5a-a7fd-27c07e72ecbb
/registrar/contact/917804081;@03dace1c3e8093de7a93d3544b4f8631: {“via_addr”:“10.10.10.4”,“qualify_timeout”:“3.000000”,“call_id”:“709681336-237”,“reg_server”:"",“prune_on_boot”:“no”,“path”:"",“endpoint”:“917804081”,“via_port”:“5070”,“authenticate_qualify”:“no”,“uri”:“sip:917804081@10.10.10.4:5070”,“qualify_frequency”:“0”,“user_agent”:“MitraStar GPT-2541GNAC ES_s00.00_g001_100VNJ0b38_2”,“expiration_time”:“1526936979”,“outbound_proxy”:""}
/registrar/contact/917804082;@eadc3dcfe2bb532d4a806de699f1b315: {“via_addr”:“10.10.10.3”,“qualify_timeout”:“3.000000”,“call_id”:“4937aa68e7e2b8ab@10.10.10.3:5070”,“reg_server”:"",“prune_on_boot”:“no”,“path”:"",“endpoint”:“917804082”,“via_port”:“5070”,“authenticate_qualify”:“no”,“uri”:“sip:917804082@10.10.10.3:5070”,“qualify_frequency”:“0”,“user_agent”:“Brcm-Cctk/v2.2.0 M5T SIP Stack/4.1.10.16”,“expiration_time”:“1526936962”,“outbound_proxy”:""}
3 results found.
*CLI>

Your contacts are definitely registered. I’m not sure why it would not be finding them. Haven’t seen anything like that, or any problems reported for that.

Man now works!!
i do a chmod 777 to astdb.sqlite

*CLI> pjsip show contacts

Contact: <Aor/ContactUri…> <Hash…> <RTT(ms)…>

Contact: 917804081/sip:917804081@10.10.10.1:5070 62e0ee1ed4 Unknown nan
Contact: 917804082/sip:917804082@10.10.10.75:2051;line= c1ae405903 Unknown nan
Contact: 917804084/sip:917804084@10.10.10.4:5070 3151bbabc4 Unknown nan

Objects found: 3

thank you so so much.

First, try this:
exten => _9[1-8]XXXXXXX,1,Dial(PJSIP/${EXTEN},25)

Check your context in dialed extension - it’s must be according the context in diaplan - I see [default].