vicidial.org

Hoping someone could enligthen me on how to enable the ODBC Storage option for Voicemail and currently using a vicibox v10 iso as an installer. I know its about asterisk installation already but I notice while on the process of installing the vicibox I didn't encounter any selection or option for the asterisk configuration, I thought I need to install it separately but as I checked it was already installed together with the vicidial. So I don't know what configurations was set on the asterisk's side. Since I'm new to using the vicibox I tried finding any documentation regarding this and to no avail. So I hope someone could help pe on this. TIA

I also encountered an app_meetme warning upon logging in as agent. I already double checked the conferences and vicidial_conferences tables and the said Conference number was present there. Maybe I also need to enable the app_meetme at asterisk but I don't know how to do it on vicibox. See below logs for reference.

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[Mar 12 10:48:22]     -- Called 55558600051@default
[Mar 12 10:48:22]     -- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-00000002;2", "8600051,K") in new stack
[Mar 12 10:48:22] WARNING[24144][C-00000005]: app_meetme.c:5261 admin_exec: Conference number '8600051' not found!
[Mar 12 10:48:22]     -- Executing [55558600051@default:2] Hangup("Local/55558600051@default-00000002;2", "") in new stack
[Mar 12 10:48:22]   == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-00000002;2'
[Mar 12 10:48:22] WARNING[24144][C-00000005]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Mar 12 10:48:22]     -- Executing [h@default:1] AGI("Local/55558600051@default-00000002;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Mar 12 10:48:22]     -- <Local/55558600051@default-00000002;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0

For whatever reason, that is normal in Asterisk when logging out and issuing a kick-all from the conference.

As for ODBC storage of Voicemail, I've never tried it before. We try to stay away from any "Asterisk ODBC" usage due to inconsistencies and problems with stability of using it at a high volume when we tested it years ago.

Thanks for the reply Matt, I had no other choice but to download the asterisk tar and run the menuselect to enable the odbc. I already tested my voicemail and its working fine and saving logs on my voicemail table. As for the conference issue, I found out that I lack the ASTconf3way screen on my keepalive. As of now, I'm currently stucked on enabling port 8089 for the webrtc. I already added the neccessary configurations on /etc/asterisk/http.conf, restarted asterisk, rebooted the vicibox and even disabled the firewall to test. I'm able to access other ports like 8088, 22, 80 and 443. I noticed that only port 8088 has an open tcp connection.

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vicibox10:~ # netstat -ant | grep 8088
tcp        0      0 0.0.0.0:8088            0.0.0.0:*               LISTEN


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vicibox10*CLI> http show status
HTTP Server Status:
Prefix:
Server: Asterisk/13.38.2
Server Enabled and Bound to 0.0.0.0:8088

Enabled URI's:
/httpstatus => Asterisk HTTP General Status
/phoneprov/... => Asterisk HTTP Phone Provisioning Tool
/ws => Asterisk HTTP WebSocket

Enabled Redirects:
  None.


Here's my config on http.conf

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/etc/asterisk/http.conf
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/root/.acme.sh//my.hostname.com/fullchain.cer
tlsprivatekey=/root/.acme.sh//my.hostname.com/vicibox10.slash.ph.key

for the 3way conf add --cu3way to your keepalives crontab entry

carpenox wrote:for the 3way conf add --cu3way to your keepalives crontab entry

Many thanks carpenox, that did the trick. I'm now stuck on enabling the tls for the websocket. I already quadrouple check all the configs as well as trying to disable firewall but same result, port 8089 still not open. Maybe this is the result on me reinstalling again the asterisk. I'm currently trying to run again ./configure with --with-ssl --enable-asteriskssl hoping it will solve it

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vicibox10:~ # netstat -tulpan| grep asterisk
tcp        0      0 0.0.0.0:5038            0.0.0.0:*               LISTEN      1013/asterisk
tcp        0      0 0.0.0.0:8088            0.0.0.0:*               LISTEN      1013/asterisk
tcp        0      0 10.100.200.109:20564    10.100.200.109:3306     ESTABLISHED 1013/asterisk
tcp        0      0 127.0.0.1:5038          127.0.0.1:59104         ESTABLISHED 1013/asterisk
tcp        0      0 127.0.0.1:5038          127.0.0.1:59102         ESTABLISHED 1013/asterisk
udp        0      0 0.0.0.0:4520            0.0.0.0:*                           1013/asterisk
udp        0      0 0.0.0.0:4569            0.0.0.0:*                           1013/asterisk
udp        0      0 0.0.0.0:2727            0.0.0.0:*                           1013/asterisk
udp        0      0 0.0.0.0:5060            0.0.0.0:*                           1013/asterisk


--with-ssl --enable-asteriskssl solved my problem on enabling tls, port 8089 is now open.

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vicibox10:~ # netstat -tulpan| grep asterisk
tcp        0      0 0.0.0.0:8088            0.0.0.0:*               LISTEN      1011/asterisk
tcp        0      0 0.0.0.0:8089            0.0.0.0:*               LISTEN      1011/asterisk
tcp        0      0 0.0.0.0:5038            0.0.0.0:*               LISTEN      1011/asterisk
tcp        0      0 10.100.200.109:42462    10.100.200.109:3306     ESTABLISHED 1011/asterisk
tcp        0      0 127.0.0.1:5038          127.0.0.1:43444         ESTABLISHED 1011/asterisk
tcp        0      0 127.0.0.1:5038          127.0.0.1:43446         ESTABLISHED 1011/asterisk
udp        0      0 0.0.0.0:4520            0.0.0.0:*                           1011/asterisk
udp        0      0 0.0.0.0:4569            0.0.0.0:*                           1011/asterisk
udp        0      0 0.0.0.0:2727            0.0.0.0:*                           1011/asterisk
udp        0      0 0.0.0.0:5060            0.0.0.0:*                           1011/asterisk


I'm now having trouble in my login prompting my sip to go busy. I'm currently checking on it and finding out what causes this. I noticed this strange IP 192.168.252.3 at the end of the logs which is not my public IP.

MyPublicIP -> The public IP of my pc where I tried to login
10.100.200.109 -> Vicibox's local IP

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[Mar 16 00:41:25]     -- Called cc107
[Mar 16 00:41:25]
[Mar 16 00:41:25] <--- SIP read from WS:MyPublicIP:5381 --->
[Mar 16 00:41:25] SIP/2.0 100 Trying
[Mar 16 00:41:25] Via: SIP/2.0/WS 10.100.200.109:5060;branch=z9hG4bK0457c0cd;rport
[Mar 16 00:41:25] From: "ACagcW1647362486777777777777" <sip:0000000000@10.100.200.109>;tag=as3735c415
[Mar 16 00:41:25] To: <sip:s9n293i9@192.0.2.159;transport=wss>
[Mar 16 00:41:25] CSeq: 102 INVITE
[Mar 16 00:41:25] Call-ID: 799b54a109251a887f8ce23a2a2a3ff0@10.100.200.109:5060
[Mar 16 00:41:25] Supported: outbound
[Mar 16 00:41:25] User-Agent: VICIphone 2.1
[Mar 16 00:41:25] Content-Length: 0
[Mar 16 00:41:25]
[Mar 16 00:41:25] <------------->
[Mar 16 00:41:25] --- (9 headers 0 lines) ---
[Mar 16 00:41:25]
[Mar 16 00:41:25] <--- SIP read from WS:MyPublicIP:5381 --->
[Mar 16 00:41:25] SIP/2.0 180 Ringing
[Mar 16 00:41:25] Via: SIP/2.0/WS 10.100.200.109:5060;branch=z9hG4bK0457c0cd;rport
[Mar 16 00:41:25] From: "ACagcW1647362486777777777777" <sip:0000000000@10.100.200.109>;tag=as3735c415
[Mar 16 00:41:25] To: <sip:s9n293i9@192.0.2.159;transport=wss>;tag=dgf3j646gf
[Mar 16 00:41:25] CSeq: 102 INVITE
[Mar 16 00:41:25] Call-ID: 799b54a109251a887f8ce23a2a2a3ff0@10.100.200.109:5060
[Mar 16 00:41:25] Supported: outbound
[Mar 16 00:41:25] User-Agent: VICIphone 2.1
[Mar 16 00:41:25] Contact: <sip:s9n293i9@192.0.2.159;transport=wss>
[Mar 16 00:41:25] Content-Length: 0
[Mar 16 00:41:25]
[Mar 16 00:41:25] <------------->
[Mar 16 00:41:25] --- (10 headers 0 lines) ---
[Mar 16 00:41:25] sip_route_dump: route/path hop: <sip:s9n293i9@192.0.2.159;transport=wss>
[Mar 16 00:41:25]     -- SIP/cc107-00000001 is ringing
[Mar 16 00:41:25]
[Mar 16 00:41:25] <--- SIP read from WS:MyPublicIP:5381 --->
[Mar 16 00:41:25] [b]SIP/2.0 480 Temporarily Unavailable[/b]
[Mar 16 00:41:25] Via: SIP/2.0/WS 10.100.200.109:5060;branch=z9hG4bK0457c0cd;rport
[Mar 16 00:41:25] From: "ACagcW1647362486777777777777" <sip:0000000000@10.100.200.109>;tag=as3735c415
[Mar 16 00:41:25] To: <sip:s9n293i9@192.0.2.159;transport=wss>;tag=dgf3j646gf
[Mar 16 00:41:25] CSeq: 102 INVITE
[Mar 16 00:41:25] Call-ID: 799b54a109251a887f8ce23a2a2a3ff0@10.100.200.109:5060
[Mar 16 00:41:25] Supported: outbound
[Mar 16 00:41:25] User-Agent: VICIphone 2.1
[Mar 16 00:41:25] Content-Length: 0
[Mar 16 00:41:25]
[Mar 16 00:41:25] <------------->
[Mar 16 00:41:25] --- (9 headers 0 lines) ---
[Mar 16 00:41:25] Transmitting (NAT) to MyPublicIP:5381:
[Mar 16 00:41:25] ACK sip:s9n293i9@192.0.2.159;transport=wss SIP/2.0
[Mar 16 00:41:25] Via: SIP/2.0/WS 10.100.200.109:5060;branch=z9hG4bK0457c0cd;rport
[Mar 16 00:41:25] Max-Forwards: 70
[Mar 16 00:41:25] From: "ACagcW1647362486777777777777" <sip:0000000000@10.100.200.109>;tag=as3735c415
[Mar 16 00:41:25] To: <sip:s9n293i9@192.0.2.159;transport=wss>;tag=dgf3j646gf
[Mar 16 00:41:25] Contact: <sip:0000000000@10.100.200.109:5060;transport=ws>
[Mar 16 00:41:25] Call-ID: 799b54a109251a887f8ce23a2a2a3ff0@10.100.200.109:5060
[Mar 16 00:41:25] CSeq: 102 ACK
[Mar 16 00:41:25] User-Agent: Asterisk PBX 13.38.2
[Mar 16 00:41:25] Content-Length: 0
[Mar 16 00:41:25]
[Mar 16 00:41:25] <------------->
[Mar 16 00:41:25]     -- SIP/cc107-00000001 is busy
[Mar 16 00:41:25] Scheduling destruction of SIP dialog '799b54a109251a887f8ce23a2a2a3ff0@10.100.200.109:5060' in 6400 ms (Method: INVITE)
[Mar 16 00:41:26]   == Manager 'sendcron' logged off from 127.0.0.1
[Mar 16 00:41:32] Reliably Transmitting (NAT) to 192.168.252.3:5060:
[Mar 16 00:41:32] OPTIONS sip:cc100@192.168.252.3:5060;rinstance=4438c5ff504dbac3;transport=UDP SIP/2.0
[Mar 16 00:41:32] Via: SIP/2.0/UDP 10.100.200.109:5060;branch=z9hG4bK47963fce;rport
[Mar 16 00:41:32] Max-Forwards: 70
[Mar 16 00:41:32] From: "asterisk" <sip:asterisk@10.100.200.109>;tag=as48e84217
[Mar 16 00:41:32] To: <sip:cc100@192.168.252.3:5060;rinstance=4438c5ff504dbac3;transport=UDP>
[Mar 16 00:41:32] Contact: <sip:asterisk@10.100.200.109:5060>
[Mar 16 00:41:32] Call-ID: 518071b74e3b218a771518a41da63388@10.100.200.109:5060
[Mar 16 00:41:32] CSeq: 102 OPTIONS
[Mar 16 00:41:32] User-Agent: Asterisk PBX 13.38.2
[Mar 16 00:41:32] Date: Tue, 15 Mar 2022 16:41:32 GMT
[Mar 16 00:41:32] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Mar 16 00:41:32] Supported: replaces, timer
[Mar 16 00:41:32] Content-Length: 0

Its working fine now. I just reinstalled the dahdi linux and dahdi-devel, rerun ./configure with dahdi, and made some adjustments on sip.conf.

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[Mar 16 20:25:58]     -- Called cc106
[Mar 16 20:25:58]
[Mar 16 20:25:58] <--- SIP read from WS:192.168.254.241:30207 --->
[Mar 16 20:25:58] SIP/2.0 100 Trying
[Mar 16 20:25:58] Via: SIP/2.0/WS 10.100.200.109:5060;branch=z9hG4bK250a7a00;rport
[Mar 16 20:25:58] From: "ACagcW1647433557777777777777" <sip:0000000000@10.100.200.109>;tag=as5ed956c3
[Mar 16 20:25:58] To: <sip:nqd6b2be@192.0.2.10;transport=wss>
[Mar 16 20:25:58] CSeq: 102 INVITE
[Mar 16 20:25:58] Call-ID: 577bb94d2044db9e2fd98f007a16dc17@10.100.200.109:5060
[Mar 16 20:25:58] Supported: outbound
[Mar 16 20:25:58] User-Agent: VICIphone 2.1
[Mar 16 20:25:58] Content-Length: 0
[Mar 16 20:25:58]
[Mar 16 20:25:58] <------------->
[Mar 16 20:25:58] --- (9 headers 0 lines) ---
[Mar 16 20:25:58]
[Mar 16 20:25:58] <--- SIP read from WS:192.168.254.241:30207 --->
[Mar 16 20:25:58] SIP/2.0 180 Ringing
[Mar 16 20:25:58] Via: SIP/2.0/WS 10.100.200.109:5060;branch=z9hG4bK250a7a00;rport
[Mar 16 20:25:58] From: "ACagcW1647433557777777777777" <sip:0000000000@10.100.200.109>;tag=as5ed956c3
[Mar 16 20:25:58] To: <sip:nqd6b2be@192.0.2.10;transport=wss>;tag=timqopco65
[Mar 16 20:25:58] CSeq: 102 INVITE
[Mar 16 20:25:58] Call-ID: 577bb94d2044db9e2fd98f007a16dc17@10.100.200.109:5060
[Mar 16 20:25:58] Supported: outbound
[Mar 16 20:25:58] User-Agent: VICIphone 2.1
[Mar 16 20:25:58] Contact: <sip:nqd6b2be@192.0.2.10;transport=wss>
[Mar 16 20:25:58] Content-Length: 0
[Mar 16 20:25:58]
[Mar 16 20:25:58] <------------->
[Mar 16 20:25:58] --- (10 headers 0 lines) ---
[Mar 16 20:25:58] sip_route_dump: route/path hop: <sip:nqd6b2be@192.0.2.10;transport=wss>
[Mar 16 20:25:58]     -- SIP/cc106-00000007 is ringing
[Mar 16 20:25:58]
[Mar 16 20:25:58] <--- SIP read from WS:192.168.254.241:30207 --->
[Mar 16 20:25:58] SIP/2.0 486 Busy Here
[Mar 16 20:25:58] Via: SIP/2.0/WS 10.100.200.109:5060;branch=z9hG4bK250a7a00;rport
[Mar 16 20:25:58] From: "ACagcW1647433557777777777777" <sip:0000000000@10.100.200.109>;tag=as5ed956c3
[Mar 16 20:25:58] To: <sip:nqd6b2be@192.0.2.10;transport=wss>;tag=timqopco65
[Mar 16 20:25:58] CSeq: 102 INVITE
[Mar 16 20:25:58] Call-ID: 577bb94d2044db9e2fd98f007a16dc17@10.100.200.109:5060
[Mar 16 20:25:58] Supported: outbound
[Mar 16 20:25:58] User-Agent: VICIphone 2.1
[Mar 16 20:25:58] Content-Length: 0
[Mar 16 20:25:58]
[Mar 16 20:25:58] <------------->
[Mar 16 20:25:58] --- (9 headers 0 lines) ---
[Mar 16 20:25:58]     -- Got SIP response 486 "Busy Here" back from 192.168.254.241:30207
[Mar 16 20:25:58] Transmitting (NAT) to 192.168.254.241:30207:
[Mar 16 20:25:58] ACK sip:nqd6b2be@192.0.2.10;transport=wss SIP/2.0
[Mar 16 20:25:58] Via: SIP/2.0/WS 10.100.200.109:5060;branch=z9hG4bK250a7a00;rport
[Mar 16 20:25:58] Max-Forwards: 70
[Mar 16 20:25:58] From: "ACagcW1647433557777777777777" <sip:0000000000@10.100.200.109>;tag=as5ed956c3
[Mar 16 20:25:58] To: <sip:nqd6b2be@192.0.2.10;transport=wss>;tag=timqopco65
[Mar 16 20:25:58] Contact: <sip:0000000000@10.100.200.109:5060;transport=ws>
[Mar 16 20:25:58] Call-ID: 577bb94d2044db9e2fd98f007a16dc17@10.100.200.109:5060
[Mar 16 20:25:58] CSeq: 102 ACK
[Mar 16 20:25:58] User-Agent: Asterisk PBX 13.38.2
[Mar 16 20:25:58] Content-Length: 0
[Mar 16 20:25:58]
[Mar 16 20:25:58]
[Mar 16 20:25:58] ---
[Mar 16 20:25:58]     -- SIP/cc106-00000007 is busy
[Mar 16 20:25:59] Really destroying SIP dialog '577bb94d2044db9e2fd98f007a16dc17@10.100.200.109:5060' Method: INVITE
[Mar 16 20:25:59]   == Manager 'sendcron' logged off from 127.0.0.1
[Mar 16 20:26:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Mar 16 20:26:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Mar 16 20:26:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Mar 16 20:26:04] Reliably Transmitting (NAT) to 192.168.254.241:30175:
[Mar 16 20:26:04] OPTIONS sip:ljo6m0ci@192.0.2.59;transport=wss SIP/2.0
[Mar 16 20:26:04] Via: SIP/2.0/WS 10.100.200.109:5060;branch=z9hG4bK6ef24190;rport
[Mar 16 20:26:04] Max-Forwards: 70
[Mar 16 20:26:04] From: "asterisk" <sip:asterisk@10.100.200.109>;tag=as514e354e
[Mar 16 20:26:04] To: <sip:ljo6m0ci@192.0.2.59;transport=wss>
[Mar 16 20:26:04] Contact: <sip:asterisk@10.100.200.109:5060;transport=ws>
[Mar 16 20:26:04] Call-ID: 5a4d0d4677ce4eaf2d12fd5702f0fad4@10.100.200.109:5060
[Mar 16 20:26:04] CSeq: 102 OPTIONS
[Mar 16 20:26:04] User-Agent: Asterisk PBX 13.38.2
[Mar 16 20:26:04] Date: Wed, 16 Mar 2022 12:26:04 GMT
[Mar 16 20:26:04] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Mar 16 20:26:04] Supported: replaces, timer
[Mar 16 20:26:04] Content-Length: 0
[Mar 16 20:26:04]
[Mar 16 20:26:04]
[Mar 16 20:26:04] ---
[Mar 16 20:26:04] ERROR[997]: chan_sip.c:4294 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
[Mar 16 20:26:04]   == Manager 'sendcron' logged off from 127.0.0.1
[Mar 16 20:26:07]   == Manager 'sendcron' logged on from 127.0.0.1
[Mar 16 20:26:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Mar 16 20:26:07]
[Mar 16 20:26:07] <--- SIP read from WS:192.168.254.241:30207 --->
[Mar 16 20:26:07] REGISTER sip:10.100.200.109 SIP/2.0
[Mar 16 20:26:07] Via: SIP/2.0/TCP 192.0.2.10;branch=z9hG4bK6328088
[Mar 16 20:26:07] To: "cc106" <sip:cc106@10.100.200.109>
[Mar 16 20:26:07] From: "cc106" <sip:cc106@10.100.200.109>;tag=cj6rmkmc3l
[Mar 16 20:26:07] CSeq: 5908 REGISTER
[Mar 16 20:26:07] Call-ID: 2j93mrd0is2n7dimdopnkv
[Mar 16 20:26:07] Max-Forwards: 70
[Mar 16 20:26:07] Authorization: Digest algorithm=MD5, username="cc106", realm="asterisk", nonce="12a134c7", uri="sip:10.100.200.109", response="f0d9537df77db5a6c2fdd22b69e4e10d"
[Mar 16 20:26:07] Contact: <sip:nqd6b2be@192.0.2.10;transport=wss>;expires=0
[Mar 16 20:26:07] Supported: outbound, path, gruu
[Mar 16 20:26:07] User-Agent: VICIphone 2.1
[Mar 16 20:26:07] Content-Length: 0
[Mar 16 20:26:07]
[Mar 16 20:26:07] <------------->
[Mar 16 20:26:07] --- (12 headers 0 lines) ---
[Mar 16 20:26:07] Sending to 192.168.254.241:30207 (NAT)
[Mar 16 20:26:07] NOTICE[21656]: chan_sip.c:17389 check_auth: Correct auth, but based on stale nonce received from '"cc106" <sip:cc106@10.100.200.109>;tag=cj6rmkmc3l'
[Mar 16 20:26:07]
[Mar 16 20:26:07] <--- Transmitting (NAT) to 192.168.254.241:30207 --->
[Mar 16 20:26:07] SIP/2.0 401 Unauthorized
[Mar 16 20:26:07] Via: SIP/2.0/TCP 192.0.2.10;branch=z9hG4bK6328088;received=192.168.254.241;rport=30207
[Mar 16 20:26:07] From: "cc106" <sip:cc106@10.100.200.109>;tag=cj6rmkmc3l
[Mar 16 20:26:07] To: "cc106" <sip:cc106@10.100.200.109>;tag=as43c237dd
[Mar 16 20:26:07] Call-ID: 2j93mrd0is2n7dimdopnkv
[Mar 16 20:26:07] CSeq: 5908 REGISTER
[Mar 16 20:26:07] Server: Asterisk PBX 13.38.2
[Mar 16 20:26:07] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Mar 16 20:26:07] Supported: replaces, timer
[Mar 16 20:26:07] WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7e31f5da", stale=true
[Mar 16 20:26:07] Content-Length: 0


LEOXXX wrote:Its working fine now. I just reinstalled the dahdi linux and dahdi-devel, rerun ./configure with dahdi, and made some adjustments on sip.conf.

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