# Can't make internal calls **URL:** \ **Category:** Asterisk SIP\ **Created:** [May 28, 2018, 3:37pm UTC](https://community.asterisk.org/t/cant-make-internal-calls/74764 "2018-05-28T15:37:56Z")\ **Posts on this page:** 14\ **Page:** 1
**Author:** ![antiwingates](https://avatars.discourse-cdn.com/v4/letter/a/d07c76/32.png) [@antiwingates](https://community.asterisk.org/u/antiwingates)\ **Post date:** [May 28, 2018, 3:37pm UTC](https://community.asterisk.org/t/cant-make-internal-calls/74764/1 "2018-05-28T15:37:56Z")
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](mailto: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](mailto: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](mailto: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](mailto: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. ---
**Author:** ![Pentium-5](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/pentium-5/32/1721_2.png) [@Pentium-5](https://community.asterisk.org/u/Pentium-5)\ **Post date:** [May 28, 2018, 4:43pm UTC](https://community.asterisk.org/t/cant-make-internal-calls/74764/2 "2018-05-28T16:43:35Z")
Accrording to the below excerpt from your configuration, 917804084 is the dynamic endpoint which should be registered on Asterisk before it can be called: ```auto [917804084] type = aor max_contacts = 1 ``` However, it looks like it is not available: ```auto == 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` ---
**Author:** ![antiwingates](https://avatars.discourse-cdn.com/v4/letter/a/d07c76/32.png) [@antiwingates](https://community.asterisk.org/u/antiwingates)\ **Post date:** [May 29, 2018, 1:49pm UTC](https://community.asterisk.org/t/cant-make-internal-calls/74764/3 "2018-05-29T13:49:20Z")
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. ---
**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\ **Post date:** [May 29, 2018, 1:50pm UTC](https://community.asterisk.org/t/cant-make-internal-calls/74764/4 "2018-05-29T13:50:34Z")
How are you storing things? Have you modified sorcery.conf and are using realtime or something else? ---
**Author:** ![antiwingates](https://avatars.discourse-cdn.com/v4/letter/a/d07c76/32.png) [@antiwingates](https://community.asterisk.org/u/antiwingates)\ **Post date:** [May 29, 2018, 1:56pm UTC](https://community.asterisk.org/t/cant-make-internal-calls/74764/5 "2018-05-29T13:56:40Z")
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 ---
**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\ **Post date:** [May 29, 2018, 2:03pm UTC](https://community.asterisk.org/t/cant-make-internal-calls/74764/6 "2018-05-29T14:03:34Z")
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. ---
**Author:** ![antiwingates](https://avatars.discourse-cdn.com/v4/letter/a/d07c76/32.png) [@antiwingates](https://community.asterisk.org/u/antiwingates)\ **Post date:** [May 29, 2018, 2:08pm UTC](https://community.asterisk.org/t/cant-make-internal-calls/74764/7 "2018-05-29T14:08:06Z")
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 ---
**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\ **Post date:** [May 29, 2018, 2:11pm UTC](https://community.asterisk.org/t/cant-make-internal-calls/74764/8 "2018-05-29T14:11:58Z")
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. ---
**Author:** ![antiwingates](https://avatars.discourse-cdn.com/v4/letter/a/d07c76/32.png) [@antiwingates](https://community.asterisk.org/u/antiwingates)\ **Post date:** [May 29, 2018, 2:28pm UTC](https://community.asterisk.org/t/cant-make-internal-calls/74764/9 "2018-05-29T14:28:42Z")
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. ---
**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\ **Post date:** [May 29, 2018, 2:30pm UTC](https://community.asterisk.org/t/cant-make-internal-calls/74764/10 "2018-05-29T14:30:40Z")
What does “database show” show? Does Asterisk have permissions to create and use /var/lib/asterisk/astdb.sqlite3? ---
**Author:** ![antiwingates](https://avatars.discourse-cdn.com/v4/letter/a/d07c76/32.png) [@antiwingates](https://community.asterisk.org/u/antiwingates)\ **Post date:** [May 29, 2018, 3:44pm UTC](https://community.asterisk.org/t/cant-make-internal-calls/74764/11 "2018-05-29T15:44:51Z")
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\> ---
**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\ **Post date:** [May 29, 2018, 3:46pm UTC](https://community.asterisk.org/t/cant-make-internal-calls/74764/12 "2018-05-29T15:46:25Z")
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. ---
**Author:** ![antiwingates](https://avatars.discourse-cdn.com/v4/letter/a/d07c76/32.png) [@antiwingates](https://community.asterisk.org/u/antiwingates)\ **Post date:** [May 29, 2018, 3:49pm UTC](https://community.asterisk.org/t/cant-make-internal-calls/74764/13 "2018-05-29T15:49:29Z")
Man now works!! i do a chmod 777 to astdb.sqlite \*CLI\> pjsip show contacts # Contact: \ \ \ 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. ---
**Author:** ![olegbo](https://avatars.discourse-cdn.com/v4/letter/o/4da419/32.png) [@olegbo](https://community.asterisk.org/u/olegbo)\ **Post date:** [June 3, 2018, 4:53am UTC](https://community.asterisk.org/t/cant-make-internal-calls/74764/14 "2018-06-03T04:53:11Z")
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].