@GarFin wrote:
Hi , I’ve installed “Dynamic Routing” in a hope of being able to forward incoming acclers whom exist in Asterisk Address book, direct to the Main Extension, rather than forcing them through the IVR.
But how to point to Asterisk Address book (or some other file/table etc) as a Source?
Regards.
Posts: 1
Participants: 1
@otaro wrote:
Hello
i need to know freepbx latest version is support sip protocal sip - i ? how to config it ?Thankyou
Posts: 2
Participants: 2
@alreilly wrote:
Folks, has anyone streamed music on hold through FreePBX from TuneIN or Pandora? I am finding that only certain URL’s will work in the streaming for FreePBX.
Posts: 1
Participants: 1
@mejor wrote:
Hey Everyone!
I am working on making when a specific extension (200) calls another (100), a 3rd (1000) is auto dialed and added to the bridge between 100 and 200. Once either 100 or 200 hang up I am needing to drop the entire 3 way call. I have tested ChanSpy app but not working either. This 3rd extension is just a paging system to broadcast the call to a room. This is what I have so far.[custom-auto-bridge]
exten => 100,1,GotoIf($[${CALLERID(num)} != 200]?nobridge)
same => n,Dial(SIP/1000,10)
same => n,GotoIf($[${DIALSTATUS} != ANSWER]?nobridge)
same => n,Bridge(${CHANNELID})
same => n(nobridge),Noop()Thoughts anyone? I really appreciate the help. I am sure this is simple and it is late and I am not seeing it. hahaha
Posts: 1
Participants: 1
@twojtczak wrote:
Hi,
how to remove this data from db to reclaim hdd space?
Posts: 5
Participants: 4
@alreilly wrote:
Greetings everyone - I am sure y’all have the same issue I have with the RoboCallers calling in. What I am looking for is that if there is a Blacklist function such as if you want to block an NPA/NXX (ex: 800555XXXX) but allow (ex: 8005551234) as there might a legitimate number that may need to get through. I know of the Blacklist Feature in FreePBX, but you have to enter each number and that could be a very long process. Does anyone have any other ideas or will FreePBX be coming out with a feature like this in the future? Thanks !
Posts: 3
Participants: 3
@sumic wrote:
hi all,
i try to use WEBRTC make a call from chrome use sipml5,
but the CLI show message “certificate verify failed”,how can i troubleshoot?
i have already have the certificate file by domain :webphoen.qinweigroup.net
thank you ~~
best regards![2019-09-09 17:40:54] ERROR[28748]: res_pjsip.c:4261 endpt_send_request: Error 171060 'Unsupported transport (PJSIP_EUNSUPTRANSPORT)' sending OPTIONS request to endpoint 6001 <--- Received SIP request (2894 bytes) from WSS:110.184.145.26:5540 ---> INVITE sip:*69@webphone.qinweigroup.net SIP/2.0 Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKS3sMI7IZ01LhtDwz4Kvl9zk3ykG1MBg9;rport From: "jack"<sip:6001@webphone.qinweigroup.net:6871>;tag=ztH6w8bTSiWBixf0nXMS To: <sip:*69@webphone.qinweigroup.net> Contact: "jack"<sips:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr" Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46 CSeq: 7204 INVITE Content-Type: application/sdp Content-Length: 2286 Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom v=0 o=- 8214160573771452000 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE 0 a=msid-semantic: WMS QLl0JSDpbxzpS9IuFk8n7GcABzsRB2pVwD80 m=audio 7306 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 110.184.145.26 a=rtcp:7307 IN IP4 110.184.145.26 a=candidate:1002299927 1 udp 2122260223 192.168.6.80 41131 typ host generation 0 network-id 1 a=candidate:1002299927 2 udp 2122260222 192.168.6.80 58302 typ host generation 0 network-id 1 a=candidate:1967005415 1 tcp 1518280447 192.168.6.80 9 typ host tcptype active generation 0 network-id 1 a=candidate:1967005415 2 tcp 1518280446 192.168.6.80 9 typ host tcptype active generation 0 network-id 1 a=candidate:3450706883 1 udp 1686052607 110.184.145.26 7306 typ srflx raddr 192.168.6.80 rport 41131 generation 0 network-id 1 a=candidate:3450706883 2 udp 1686052606 110.184.145.26 7307 typ srflx raddr 192.168.6.80 rport 58302 generation 0 network-id 1 a=ice-ufrag:/D80 a=ice-pwd:5okDbtNyoE6yUkMzLXBiqlkN a=ice-options:trickle a=fingerprint:sha-256 A9:32:22:20:31:BE:33:67:6A:15:A4:E7:51:96:CA:56:B6:8B:8B:7C:92:5A:95:1B:FE:26:CB:47:94:AC:57:ED a=setup:actpass a=mid:0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv a=msid:QLl0JSDpbxzpS9IuFk8n7GcABzsRB2pVwD80 d4197ebc-3f1d-4476-bd11-b6b57191d46c a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:110 telephone-event/48000 a=rtpmap:112 telephone-event/32000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 a=ssrc:2150700876 cname:vScPyFZzs73nSNMx a=ssrc:2150700876 msid:QLl0JSDpbxzpS9IuFk8n7GcABzsRB2pVwD80 d4197ebc-3f1d-4476-bd11-b6b57191d46c a=ssrc:2150700876 mslabel:QLl0JSDpbxzpS9IuFk8n7GcABzsRB2pVwD80 a=ssrc:2150700876 label:d4197ebc-3f1d-4476-bd11-b6b57191d46c <--- Transmitting SIP response (570 bytes) to WSS:110.184.145.26:5540 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=5540;received=110.184.145.26;branch=z9hG4bKS3sMI7IZ01LhtDwz4Kvl9zk3ykG1MBg9 Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46 From: "jack" <sip:6001@webphone.qinweigroup.net>;tag=ztH6w8bTSiWBixf0nXMS To: <sip:*69@webphone.qinweigroup.net>;tag=z9hG4bKS3sMI7IZ01LhtDwz4Kvl9zk3ykG1MBg9 CSeq: 7204 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1568022094/b5d393d275f74b8be8e01efbb6663c72",opaque="7b46d5942c5a6099",algorithm=md5,qop="auth" Server: FPBX-15.0.16(16.5.0) Content-Length: 0 <--- Received SIP request (403 bytes) from WSS:110.184.145.26:5540 ---> ACK sip:*69@webphone.qinweigroup.net SIP/2.0 Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKS3sMI7IZ01LhtDwz4Kvl9zk3ykG1MBg9;rport From: "jack"<sip:6001@webphone.qinweigroup.net:6871>;tag=ztH6w8bTSiWBixf0nXMS To: <sip:*69@webphone.qinweigroup.net>;tag=z9hG4bKS3sMI7IZ01LhtDwz4Kvl9zk3ykG1MBg9 Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46 CSeq: 7204 ACK Content-Length: 0 Max-Forwards: 70 <--- Received SIP request (3188 bytes) from WSS:110.184.145.26:5540 ---> INVITE sip:*69@webphone.qinweigroup.net SIP/2.0 Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK57egFTiLQXcqO4Ocrg7TyDE8NPbNeR8N;rport From: "jack"<sip:6001@webphone.qinweigroup.net:6871>;tag=ztH6w8bTSiWBixf0nXMS To: <sip:*69@webphone.qinweigroup.net> Contact: "jack"<sips:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr" Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46 CSeq: 7205 INVITE Content-Type: application/sdp Content-Length: 2286 Max-Forwards: 70 Authorization: Digest username="6001",realm="asterisk",nonce="1568022094/b5d393d275f74b8be8e01efbb6663c72",uri="sip:*69@webphone.qinweigroup.net",response="cb9e1bef0f3f6eb3dd7758a2aa34522c",algorithm=md5,cnonce="5147d96682bd6403b00648ffa4d0e929",opaque="7b46d5942c5a6099",qop=auth,nc=00000001 User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom v=0 o=- 8214160573771452000 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE 0 a=msid-semantic: WMS QLl0JSDpbxzpS9IuFk8n7GcABzsRB2pVwD80 m=audio 7306 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 110.184.145.26 a=rtcp:7307 IN IP4 110.184.145.26 a=candidate:1002299927 1 udp 2122260223 192.168.6.80 41131 typ host generation 0 network-id 1 a=candidate:1002299927 2 udp 2122260222 192.168.6.80 58302 typ host generation 0 network-id 1 a=candidate:1967005415 1 tcp 1518280447 192.168.6.80 9 typ host tcptype active generation 0 network-id 1 a=candidate:1967005415 2 tcp 1518280446 192.168.6.80 9 typ host tcptype active generation 0 network-id 1 a=candidate:3450706883 1 udp 1686052607 110.184.145.26 7306 typ srflx raddr 192.168.6.80 rport 41131 generation 0 network-id 1 a=candidate:3450706883 2 udp 1686052606 110.184.145.26 7307 typ srflx raddr 192.168.6.80 rport 58302 generation 0 network-id 1 a=ice-ufrag:/D80 a=ice-pwd:5okDbtNyoE6yUkMzLXBiqlkN a=ice-options:trickle a=fingerprint:sha-256 A9:32:22:20:31:BE:33:67:6A:15:A4:E7:51:96:CA:56:B6:8B:8B:7C:92:5A:95:1B:FE:26:CB:47:94:AC:57:ED a=setup:actpass a=mid:0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv a=msid:QLl0JSDpbxzpS9IuFk8n7GcABzsRB2pVwD80 d4197ebc-3f1d-4476-bd11-b6b57191d46c a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:110 telephone-event/48000 a=rtpmap:112 telephone-event/32000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 a=ssrc:2150700876 cname:vScPyFZzs73nSNMx a=ssrc:2150700876 msid:QLl0JSDpbxzpS9IuFk8n7GcABzsRB2pVwD80 d4197ebc-3f1d-4476-bd11-b6b57191d46c a=ssrc:2150700876 mslabel:QLl0JSDpbxzpS9IuFk8n7GcABzsRB2pVwD80 a=ssrc:2150700876 label:d4197ebc-3f1d-4476-bd11-b6b57191d46c == Setting global variable 'SIPDOMAIN' to 'webphone.qinweigroup.net' <--- Transmitting SIP response (374 bytes) to WSS:110.184.145.26:5540 ---> SIP/2.0 100 Trying Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=5540;received=110.184.145.26;branch=z9hG4bK57egFTiLQXcqO4Ocrg7TyDE8NPbNeR8N Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46 From: "jack" <sip:6001@webphone.qinweigroup.net>;tag=ztH6w8bTSiWBixf0nXMS To: <sip:*69@webphone.qinweigroup.net> CSeq: 7205 INVITE Server: FPBX-15.0.16(16.5.0) Content-Length: 0 == Using SIP RTP Audio TOS bits 184 == Using SIP RTP Audio TOS bits 184 in TCLASS field. == Using SIP RTP Audio CoS mark 5 -- Executing [*69@from-internal:1] Goto("PJSIP/6001-0000002d", "app-calltrace-perform,s,1") in new stack -- Goto (app-calltrace-perform,s,1) -- Executing [s@app-calltrace-perform:1] Set("PJSIP/6001-0000002d", "CONNECTEDLINE(name-charset,i)=utf8") in new stack -- Executing [s@app-calltrace-perform:2] Set("PJSIP/6001-0000002d", "CONNECTEDLINE(name,i)=呼叫追踪") in new stack -- Executing [s@app-calltrace-perform:3] Set("PJSIP/6001-0000002d", "CONNECTEDLINE(num,i)=s") in new stack -- Executing [s@app-calltrace-perform:4] Answer("PJSIP/6001-0000002d", "") in new stack <--- Transmitting SIP response (1514 bytes) to WSS:110.184.145.26:5540 ---> SIP/2.0 200 OK Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=5540;received=110.184.145.26;branch=z9hG4bK57egFTiLQXcqO4Ocrg7TyDE8NPbNeR8N Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46 From: "jack" <sip:6001@webphone.qinweigroup.net>;tag=ztH6w8bTSiWBixf0nXMS To: <sip:*69@webphone.qinweigroup.net>;tag=79c682b4-51df-4333-a663-95ff01efde84 CSeq: 7205 INVITE Server: FPBX-15.0.16(16.5.0) Contact: <sips:192.168.1.158:8089;transport=ws> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub P-Asserted-Identity: "呼叫追踪" <sip:s@webphone.qinweigroup.net> Content-Type: application/sdp Content-Length: 791 v=0 o=- 2007666272 4 IN IP4 192.168.1.158 s=Asterisk c=IN IP4 192.168.1.158 t=0 0 a=group:BUNDLE 0 m=audio 12072 UDP/TLS/RTP/SAVPF 111 8 126 a=connection:new a=setup:active a=fingerprint:SHA-256 53:49:22:7D:B4:4E:69:5F:62:F6:18:50:65:84:D0:77:EC:6E:8D:DE:AB:00:A7:76:2D:25:89:A7:77:2D:E6:4F a=ice-ufrag:23d30b5b2dbde9b75fe38a4b654da6d3 a=ice-pwd:0039026f23c975ca23d5a814280ca8af a=candidate:He0ec0f98 1 UDP 2130706431 fe80::f816:3eff:fe20:b27f 12072 typ host a=candidate:Hc0a8019e 1 UDP 2130706431 192.168.1.158 12072 typ host a=rtpmap:111 opus/48000/2 a=fmtp:111 useinbandfec=1 a=rtpmap:8 PCMA/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:126 0-16 a=ptime:20 a=maxptime:20 a=sendrecv a=rtcp-mux a=ssrc:1560523773 cname:45b90ca4-4336-40eb-a066-08032a3e4f91 a=mid:0 <--- Received SIP request (899 bytes) from WSS:110.184.145.26:5540 ---> ACK sips:192.168.1.158:8089;transport=ws SIP/2.0 Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKb5No2TLObME2ZeTA6mZu;rport From: "jack"<sip:6001@webphone.qinweigroup.net:6871>;tag=ztH6w8bTSiWBixf0nXMS To: <sip:*69@webphone.qinweigroup.net>;tag=79c682b4-51df-4333-a663-95ff01efde84 Contact: "jack"<sips:6001@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr" Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46 CSeq: 7205 ACK Content-Length: 0 Max-Forwards: 70 Authorization: Digest username="6001",realm="asterisk",nonce="1568022094/b5d393d275f74b8be8e01efbb6663c72",uri="sips:192.168.1.158:8089;transport=ws",response="f29731cd3b7930c85f64e989a7f4ed8c",algorithm=md5,cnonce="5147d96682bd6403b00648ffa4d0e929",opaque="7b46d5942c5a6099",qop=auth,nc=00000002 User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom [2019-09-09 17:41:34] ERROR[30514][C-00000023]: res_rtp_asterisk.c:2970 __rtp_recvfrom: DTLS failure occurred on RTP instance '0x7f780408e7f0' due to reason 'certificate verify failed', terminating [2019-09-09 17:41:34] WARNING[30514][C-00000023]: res_rtp_asterisk.c:7108 ast_rtp_read: RTP Read error: Unspecified. Hanging up. == Spawn extension (app-calltrace-perform, s, 4) exited non-zero on 'PJSIP/6001-0000002d' <--- Transmitting SIP request (507 bytes) to WSS:110.184.145.26:5540 ---> BYE sips:6001@110.184.145.26:5540;transport=ws;rtcweb-breaker=no;click2call=no SIP/2.0 Via: SIP/2.0/WSS 192.168.1.158:8089;rport;branch=z9hG4bKPjd8baabe6-cc14-43de-9a61-d5efd783537e;alias From: <sip:*69@webphone.qinweigroup.net>;tag=79c682b4-51df-4333-a663-95ff01efde84 To: "jack" <sip:6001@webphone.qinweigroup.net>;tag=ztH6w8bTSiWBixf0nXMS Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46 CSeq: 14366 BYE Reason: Q.850;cause=0 Max-Forwards: 70 User-Agent: FPBX-15.0.16(16.5.0) Content-Length: 0 <--- Received SIP response (420 bytes) from WSS:110.184.145.26:5540 ---> SIP/2.0 200 OK Via: SIP/2.0/WSS 192.168.1.158:8089;rport=8089;branch=z9hG4bKPjd8baabe6-cc14-43de-9a61-d5efd783537e;alias From: <sip:*69@webphone.qinweigroup.net>;tag=79c682b4-51df-4333-a663-95ff01efde84 To: "jack"<sip:6001@webphone.qinweigroup.net>;tag=ztH6w8bTSiWBixf0nXMS Contact: <sips:6001@df7jal23ls0d.invalid;transport=wss> Call-ID: 462943b5-4f97-4506-ef3b-ddbff69a9b46 CSeq: 14366 BYE Content-Length: 0
Posts: 1
Participants: 1
@SBaalman wrote:
The GUI is super slooow logging in, pulling up the dashboard and going to other functions within the GUI.
I have added the IP and hostname to the DNS server which helped a bit for some things, the first entry in DNS setting on the unit is 127.0.0.1
Here’s what I pulled from running fwconsole debug
[root@FreePBX ~]# fwconsole debug
±----------------------+
| FreePBX Notifications |
±----------------------+
OUT > ==> /var/log/asterisk/freepbx_dbug <====> /var/log/httpd/error_log <==
140466771507088:error:0200100D:system library:fopen:Permission denied:bss_file.c:402:fopen(’/etc/pki/tls/certs/localhost.crt’,‘r’)
140466771507088:error:20074002:BIO routines:FILE_CTRL:system lib:bss_file.c:404:
unable to load certificate
[Thu Aug 29 18:38:55.443293 2019] [mpm_prefork:notice] [pid 1230] AH00170: caught SIGWINCH, shutting down gracefully
[Thu Aug 29 18:38:56.443693 2019] [core:notice] [pid 1230] AH00052: child pid 12117 exit signal Segmentation fault (11)
[Thu Aug 29 19:42:34.704777 2019] [suexec:notice] [pid 1324] AH01232: suEXEC mechanism enabled (wrapper: /usr/sbin/suexec)
[Thu Aug 29 19:42:34.772183 2019] [auth_digest:notice] [pid 1324] AH01757: generating secret for digest authentication …
[Thu Aug 29 19:42:34.772849 2019] [lbmethod_heartbeat:notice] [pid 1324] AH02282: No slotmem from mod_heartmonitor
[Thu Aug 29 19:42:35.735719 2019] [mpm_prefork:notice] [pid 1324] AH00163: Apache/2.4.6 (Sangoma) OpenSSL/1.0.2k-fips PHP/5.6.40 configured – resuming normal operations
[Thu Aug 29 19:42:35.735750 2019] [core:notice] [pid 1324] AH00094: Command line: ‘/usr/sbin/httpd -D FOREGROUND’==> /var/log/asterisk/freepbx_security.log <==
[2019-08-29 15:08:32] Authentication failure for vsradmin from 192.168.2.11
[2019-08-29 15:08:32] Possible proxy detected, forwarded headers forvsradmin set to==> /var/log/asterisk/freepbx.log <==
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module queuestats, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module qxact_reports, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module recording_report, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module restapps, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module sangomacrm, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module sipstation, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module soundlang, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module userman, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module vqplus, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
[2019-Aug-29 21:13:41] [freepbx.INFO]: Deprecated way to add Console commands for module zulu, adding console commands this way can have negative performance impacts. Please use module.xml. See: https://wiki.freepbx.org/display/FOP/Adding+fwconsole+commands [] []
OUT > [2019-Aug-29 21:16:52] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 67
OUT > [2019-Aug-29 21:16:52] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 76
OUT > [2019-Aug-29 21:16:53] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 67
[2019-Aug-29 21:16:53] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 76
OUT > [2019-Aug-29 21:18:13] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 67
OUT > [2019-Aug-29 21:18:13] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 76Here’s what is in functions.inc.php between lines 67 and 76 since those two are referenced
67 $um = module_getinfo('userman', MODULE_STATUS_ENABLED); 68 if(file_exists($amp_conf['AMPWEBROOT'].'/admin/modules/userman/functions.inc.php') && (isset($um['userman']['status']) && $um['userman']['status'] === MODULE_STATUS_ENABLED)) { 69 include_once($amp_conf['AMPWEBROOT'].'/admin/modules/userman/functions.inc.php'); 70 } 71 } 72 73 //Ensure that the manager module has loaded. If not, load it. 74 if(!function_exists('manager_add')){ 75 global $amp_conf; 76 $um = module_getinfo('manager', MODULE_STATUS_ENABLED);I noticed in debug it throws those bits about lines 67 and 76 and suddenly the GUI responds
Any ideas?
Posts: 11
Participants: 2
@duli wrote:
Hello, Forum:
I`d like to develop a special button in my company´s groupware (web based) to enable / disable call forward (*72 / *73) on the user´s extension. I know the UCP has this ability, but I´d like to provide the same functionality from inside the groupware. Any pointers or tips about where I should begin to look?
Thanks a lot!
Posts: 1
Participants: 1
@v70ff wrote:
Any “.conf” file that i try to access I get the permissions denied message.
I am trying to access it as root user “[root@freepbx ~]#”
Does anyone know why this is happening?
Posts: 9
Participants: 2
@bhoover wrote:
I had an install of FreePBX that was working fine. I came back after the weekend and now the web GUI is no longer working properly. I can log into it and the header and footers display along with the dropdowns but nothing else displays. For example, when i click on the ‘Dashboard’ link, it appears that it is loading the page but nothing displays. The same goes for all of the other links in the dropdown menus. The only error I see is ni the weserver logs: AH01630: client denied by server configuration: /var/www/html/admin/index.html
Posts: 3
Participants: 2
@dan_ce wrote:
Hi Lorne - this one I think might be a question you might be best placed to answer? It’s re: this thread where you fixed the module
Just got a spam call, and wondered why Caller ID superfecta didn’t “protect us” from it, given that when I run the number through Caller ID superfecta test, it’s shown to be a SPAM caller. It should have gone to Lenny, I thought.
When looking at the inbound route it’s set to send calls through Caller ID superfecta and indeed this did happen as the caller ID geographic region was set to ‘LONDON’ which is one of my lookup sources. However, this lookup source is BELOW the lookup source which would have flagged the call as spam?
My logs are here and as you can see at about line 62 Superfecta sets the caller ID to London.
In the demo, the caller ID would be set to SPAM, but it looks like in reality this may not happen??
Thank you!!
(Forgive me if I’ve got something mixed up)
Posts: 2
Participants: 2
@avayax wrote:
Fail2ban blocked a local IP address yesterday, which belongs to an ordinary workstation because of failed “attempts against apache-auth”.
Httpd acces logs show this.
Do those logs show that something is trying to steal my http provisioning credentials and has someone else seen thisGET /mnt/mtd/AVAST-HNS-SCAN-PROBE HTTP/1.1before?10.1.10.119 - user [08/Sep/2019:21:40:33 -0400] "GET / HTTP/1.1" 401 477 "-" "-" 10.1.10.119 - user [08/Sep/2019:21:40:33 -0400] "GET / HTTP/1.1" 401 477 "-" "-" 10.1.10.119 - user [08/Sep/2019:21:40:33 -0400] "GET / HTTP/1.1" 401 477 "-" "-" 10.1.10.119 - user [08/Sep/2019:21:40:33 -0400] "GET / HTTP/1.1" 401 477 "-" "-" 10.1.10.119 - webadmin [08/Sep/2019:21:40:33 -0400] "GET / HTTP/1.1" 401 477 "-" "-" 10.1.10.119 - webadmin [08/Sep/2019:21:40:33 -0400] "GET / HTTP/1.1" 401 477 "-" "-" 10.1.10.119 - webadmin [08/Sep/2019:21:40:34 -0400] "GET / HTTP/1.1" 401 477 "-" "-" 10.1.10.119 - webadmin [08/Sep/2019:21:40:34 -0400] "GET / HTTP/1.1" 401 477 "-" "-" 10.1.10.119 - - [08/Sep/2019:21:40:34 -0400] "GET /etc/passwd HTTP/1.1" 401 477 "-" "-" 10.1.10.119 - - [08/Sep/2019:21:40:34 -0400] "GET /language/Swedish${IFS}&&ping$IFS-c1$IFS-s41${IFS}10.1.10.119>/dev/null&&tar${IFS}/string.js HTTP/1.1" 404 384 "-" "-" 10.1.10.119 - - [08/Sep/2019:21:40:34 -0400] "GET /language/Swedish${IFS}&&echo${IFS}AVAST-HNS-SCAN-PROBE>AVAST-HNS-SCAN-PROBE&&tar${IFS}/string.js HTTP/1.1" 404 389 "-" "-" 10.1.10.119 - - [08/Sep/2019:21:40:34 -0400] "GET /mnt/mtd/AVAST-HNS-SCAN-PROBE HTTP/1.1" 404 302 "-" "-" 10.1.10.119 - - [08/Sep/2019:21:40:34 -0400] "GET /etc/passwd HTTP/1.1" 404 284 "-" "-"
Posts: 4
Participants: 3
@aratel wrote:
Hello,
I receive mails with with a storage notification :Storage space is getting critically high on the following drives of your system: /dev/sda1 is 100% fullBUT… I have more than 60 Freepbx servers running in cloud. How can I know wich one this message concerns ?
Thanks you
Posts: 1
Participants: 1
@cloudpbxfuzz wrote:
On a new FreePBX14 installation, at times (seems to be random), I cannot access the GUI . When checking the apache error_log file in /var/log/httpd, I see the line:
[Mon Sep 09 09:29:15.002509 2019] [authz_core:error] [pid 21416] [client X.X.X.X:49152] AH01630: client denied by server configuration: /var/www/html/admin/index.html
If I open an incognito window in Chrome, I can usually get to the GUI properly. Then, at other times, I can access the GUI perfectly fine using a non-incognito window.
This is a stock distro install of FPBX14. I’ve seen some other posts on here about this, but no real resolution. Has anyone else experienced this?
Posts: 1
Participants: 1
@nickb wrote:
I’m running FreePBX 12.7.6-1904-1.sng7 using the commercial EPM, on a system with 73 extensions. Most of these (about 65) are managed by the EPM; a few unsupported devices are manually configured. The system is fully updated in Yum and Module Admin.
Nearly all devices are Cisco SPA500 series. Recently we added six Grandstream GXP-1628 devices, configured via commercial EPM.
These devices successfully provision (verified in apache log) and have the correct information in the provisioning files for SIP server and ports. I am using PJSIP.
All six devices will only show “Unavailable” in SIP Peers. Attempting to dial one of these extensions from another results in the call going directly to voicemail. They can, however, dial voicemail, receive a password prompt from the server, subsequently interacting with the voicemail subsystem.
Suspecting a NAT issue in the router at the client site, we verified SIP ALG was disabled, and changed NAT settings to the recommended values for a SonicWall per the documentation that the site network admin referenced (I do not have that documentation). The UDP NAT timeout is 30 seconds in the router, so I changed Register Expiration to 1 minute, and ReRegister Before Expiration to 40 seconds in the Basefile Editor for the Grandstream template, this should result in re-registration before the 30 second UDP NAT timeout happens.
However, not even the initial registration is received when the phones are booted. I did reboot the FreePBX server as well, but no change in behavior occured after rebooting.
I’ve also verified that the site IP address is in the “Local” networks group in the Firewall, and is whitelisted in Intrusion Detection.
Any suggestions as to why the phones may not be registering at all?
Posts: 2
Participants: 2
@BGM wrote:
I like to keep a bunch of FreePBX pages open in my browser; sometimes when I return to one of those pages, the page is replaced with:
Potential Security Breach You are attempting to modify settings from a URL that does not appear to have come from a FreePBX page link or button. This can occur if you manually typed in the URL below. This action has been blocked because the HTTP_REFERER does not match your current SERVER. If you require this access, you can set Check Server Referrer=false in Advanced Settings to disable this security check The suspect URL is listed below. If this action is intended, you can click this link and your action will be processed. Do not proceed with this if you did not intended to execute this command as it may result in changes to your configuration.Is there a way to disable this? Refreshing the page doesn’t work and I have to navigate back to where I was.
Posts: 4
Participants: 2
@chasemixon wrote:
Every Inbound Route gives me this message, whenever I click submit,
Whoops\Exception\ErrorException thrown with message “Cannot unset string offsets”
Stacktrace:
#0 Whoops\Exception\ErrorException in /var/www/html/admin/modules/zulu/Zulu.class.php:1210everything seems to work, just scares me the first time I see it…
Posts: 1
Participants: 1
@v70ff wrote:
So the situations is, When a call is coming in for some reason that same call keeps jumping in and out. To answer the call you would need the perfect timing of picking up, sometimes the receptionist wold try picking up the call 3 times before she successfully does it.
Posts: 1
Participants: 1
@berjanschroer wrote:
Hi Everyone,
Currently i’m using FreePBX 14. when we have 1 voicemail in our inbox we are finding some troubles.
when we call *98XXX (or *97) we hear the greeting: You have 1 { call hangs up}.In my research on the internet i found the following issue: https://issues.freepbx.org/browse/FREEPBX-17153
This guy has the same problem that we have, when you read the article the problem has been found!my million dollar question is, how can i solve this problem? where can i copy the sound file and renaming it to fit on the missing one?
Thank in advance,
Posts: 7
Participants: 4
