Sandbox Project

[carpenox]

What exactly is the sandbox project? I noticed it uses a newer version of asterisk, 13.32.0 and was curious if this version of asterisk would work if its upgraded from a regular vicibox 9.0.2 install? what would be the drawbacks from trying it? is it compatible? or is that what sandbox is for? to test those kind of things?

[Kumba]

It's my dev sandbox. If I am building or testing new versions of packages I build them here then test them against ViciBox. So 13.32.0 is built there so we can do some performance testing, see if it crashes, etc. It's not really for production use. Think of it as the beta test area for packages.

[carpenox]

oh thats cool, I installed it on a virtual server to see how it went. I guess theres some db schema issues with it and the astgui so far correct?

[Kumba]

As far as I know it should more or less work. I basically leave it there while some people test it and roll it out if it passes.

[carpenox]

the thing i noticed so far was with adding chat user groups, it responds with an error that does not apply, and for lead loading as well. I submitted it to mantis with photos.

[carpenox]

Is there a perl script i can run with --debugx to watch as the db processes requests?

[Kumba]

Not to watch DB processes. A lot of the scripts if they see --debugx will output the actual SQL that's being ran. It's just going to depend on the processes you're running.

Submitting what you found to Mantis for something inside of ViciDial like the user group chat is the best way to approach that. If it's something outside of ViciDial like the VB-firewall.pl script or something during the install then the forums here are probably the best bet.

I'm currently going through and creating vicibox-<package> for each major part of ViciBox. Currently I've gotten vicibox-dynportal done for the dynamic portal. I'm probably going to have vicibox-firewall finished up here shortly. I'm currently planning to do one for the certbot stuff and then a sort of master 'default' vicibox-install one. Generally if the feature isn't big enough to exist in it's own orbit, it gets thrown into vicibox-install. A lot of the system dependencies will be in vicibox-install so hopefully this will help increase consistency between version. The real goal is to make ViciBox releases more OS dependent then feature/package dependent. That way if I need to adjust things like kernel boot options or ulimits or tune the OS then that's a ViciBox release. Everything else is just a zypper package update that requires less user intervention then checking a forum, hopefully.

That's the plan anyways. To basically start managing ViciBox in a more maintainable way for everyone. The really cool thing is I'm building this on the Open Build Service. This allows me to spin off variants like one for Ubuntu/Debian and one for CentOS/Fedora or even different arch's like Arm. The cross-platform work is going to require someone to figure out how to do that though. I don't have any desire currently to figure that out myself so I leave the invitation open to anyone who wants to jump in. You can create a login at OBS, clone my package, do your testing and changes, and then submit them back to me so I can see what needs to be done. It's basically an SVN/GIT method of doing actual packages and works nicely. It's how I've submitted updates and bug fixes that we in this forum have found in ViciBox up to OpenSuSE in the past.

[carpenox]

Kumba,

That is so awesome, that should help relieve some of the same repetitive questions im sure you guys see on here all the time, is there any way I can test the dynportal installation on one of my servers that doesnt use it yet? one of the scratch installs ive done. Is there a perl/bash script i can run or anything?

[Kumba]

you would add the repo then install vicibox-dynportal. This is all still mostly in test right now, but you can add it by doing the following:

Code: Select all

zypper ar https://download.opensuse.org/repositories/home:/vicidial:/vicibox/openSUSE_Leap_15.1/home:vicidial:vicibox.repo
zypper in vicibox-dynportal


None of the other packages in that repository are ready yet. I did finish the vicibox-firewall one as well but I need to test it installing it on a live system still. The package built and everything seems to be where it's supposed to be however.

[carpenox]

good stuff, i have the repo instaled on all my servers already so ill just install the package, thx ill let u know how it goes

[carpenox]

It worked like a charm, dynportal is worknig 100% for me now, thx

[carpenox]

Kumba,

Have you created a vicibox-cluster tool yet? something that could be run on existing servers to go thru the vicibox-install expert mode process? The reason it would be nice to have would be for people that have older versions or that have done scratch installs and dont have that script on our servers already. Just food for thought if you havent done it already or thought about doing it.

-Nox

[williamconley]

Have you created a vicibox-cluster tool yet? something that could be run on existing servers to go thru the vicibox-install expert mode process? The reason it would be nice to have would be for people that have older versions or that have done scratch installs and dont have that script on our servers already. Just food for thought if you havent done it already or thought about doing it.

Vicibox is proprietary. Primarily because the .iso building tools are so complex and ever changing that releasing their source would be cumbersome. But also because copycats would otherwise try to deploy confusing knockoffs that could confuse the consumer (Goautodial was once quite difficult to understand in relation to Vicidial, for instance).

So if you want a scratch install that allows for clustering, you are still (for the last decade or so) stuck with the Vicidial Multi-Server Manual (written by PoundTeam). AFAIK: Still valid. Not simplistic. Not a script. But then again not all that difficult, either.

If you want to upgrade ... there are already upgrade instructions in the SVN tree. The name of the instruction page is UPGRADE. It contains instructions to upgrade an existing server to get to the SVN revision you acquired from the svn server which contains that instruction set. Works very well. BUT: will not upgrade the OS, Asterisk, MySQL or anything outside the Vicidial scripts and database structure. For that ... you must re-install the server with a later installation .iso to get the more recent applications.

[carpenox]

what i was thinking about the "cluster tool" would be something that asks a set of questions like in the expert install that would in turn deactivate or activate certain aspects of an already existing all-in-one server... like when it comes to "will this be a database server?" and you answer yes, it would leave that part alone, then the next question "web server?" and you say no, it would then disable or output at the end what files you would need to edit in order to turn off everything you responded "no" to. And then this could be run on all servers in the cluster and optimized accordingly.

[williamconley]

This is why we install all roles in all servers. So we can re-purpose a server without having to reinstall any software. Perhaps activating/deactivating a service, but no actuall installation.

[Kumba]

There's too many variations between branches of a distro let alone distro's themselves. For instance OpenSuSE v.15.1 uses firewalld which previous version of OpenSuSE does not. This means that how the ViciBox Firewall integrates between the two is completely different. It gets even worse when you go between distros themselves like Debian or CentOS. Things like the install paths, package naming conventions (pre-requisites), boot/startup procedures, etc, all change or have 'subtleties' that problems. These could range from a feature not working correctly to a service just refusing to start. We also require a custom-patched version of Asterisk to get some features as well as packages that just don't exist in distros. So just to get to the same level as ViciBox in regards to the pre-requisites would a little a bit of work.

Making something like a vicibox-installer, which is basically a cluster setup tool, that works across various distros would really need to be it's own undertaking outside the immediate scope of ViciBox. The primary goal of ViciBox is to provide a, hopefully, well optimized and working install right out of the box. Not everyone who uses ViciBox wants to do a deep dive into the OS to figure out why things aren't working right. For instance, the default configs shipped with Apache and MySQL are not good for more then a couple dozen agent on pretty much every distro I've seen.

I'm not opposed to the idea of including more cross-platform stuff, but it's something that someone else would need to crontribute their time to at the moment. My goal right now is to make ViciBox more maintainable through the packages first. After that I can consider other expansions for ViciBox.

[carpenox]

sounds good, it was just an idea and not really even cross platform, even within the "current" build guidelines such as with vicibox 9.0.2 using opensuse 15.1. in a situation where its not technically vicibox since it wasnt installed using the .iso installer, however all aspects of the software running on those servers are essentially identical(scratch install) the "vicibox-install" would not be available to fine tune each server to particular roles such as db, telephony, etc. I guess i was looking for an easy method to do this after using the hard way to get everything worknig to begin with which is kind of a cop out. I will just have to do the research and modify the startup(rc.local) and crontab to only use what is needed for that servers role(s). thanks anyways

[williamconley]

If the goal is "easy reassignment of role(s)", then you may want to consider that at present all these settings are in .conf files. Those .conf file settings could be stored in a table record or spare .conf file location and a trigger event could cause them to be written to their respective live files. Then a simple reboot and the server is now reassigned to the new role.

As long as all required packages are installed in all servers, that would work nicely.

[carpenox]

good idea Bill, thx

[carpenox]

Kumba,

Hows 13.32.0 coming along i seen u added it to the sandbox, i tried it out but not from the sandbox, just from the asterisk site, caused a few problems so i reverted, but im curious how its going with it. ANd vici doesnt use postgres already does it?

-Nox

[williamconley]

vici doesnt use postgres already does it?

Postgres can't keep up with Vicidial servers. Only the MyIsam engine in MySQL can keep up with a vicidial server at present. Nothing else comes close.

[Kumba]

Kumba,

Hows 13.32.0 coming along i seen u added it to the sandbox, i tried it out but not from the sandbox, just from the asterisk site, caused a few problems so i reverted, but im curious how its going with it. ANd vici doesnt use postgres already does it?

-Nox

Haven't heard any complaints. The version in the sandbox has the vicidial patches so you should be able to install it without issue. Give it a shot and let me know if you ran into any weirdness.

[carpenox]

ok cool, ill keep u up to date


so far everything looks good, after restarting asterisk i dont see any errors:

Code: Select all

 *CLI> Connected to Asterisk 13.32.0-vici currently running on ip                                                                                 -

 Asterisk Queue Logger restarted
    -- Reloading module 'res_statsd.so' (StatsD client support)
    -- Reloading module 'res_pjproject.so' (PJPROJECT Log and Utility Support)
    -- Reloading module 'res_pjsip.so' (Basic SIP resource)
[May 19 23:34:31] NOTICE[23750]: sorcery.c:1333 sorcery_object_load: Type 'syste                                                                                 m' is not reloadable, maintaining previous values
    -- Reloading module 'res_crypto.so' (Cryptographic Digital Signatures)
[May 19 23:34:31] WARNING[22916]: res_crypto.c:515 crypto_load: Unable to open k                                                                                 ey directory '/usr/share/asterisk/keys'
    -- Reloading module 'res_phoneprov.so' (HTTP Phone Provisioning)
[May 19 23:34:31] WARNING[22916]: res_phoneprov.c:1233 get_defaults: Unable to f                                                                                 ind a valid server address or name.
    -- Reloading module 'res_xmpp.so' (Asterisk XMPP Interface)
    -- Reloading module 'res_stun_monitor.so' (STUN Network Monitor)
    -- Reloading module 'res_pjsip_outbound_publish.so' (PJSIP Outbound Publish                                                                                  Support)
    -- Reloading module 'res_ari.so' (Asterisk RESTful Interface)
[May 19 23:34:31] ERROR[22916]: ari/config.c:312 process_config: No configured u                                                                                 sers for ARI
    -- Reloading module 'res_fax.so' (Generic FAX Applications)
    -- Reloading module 'res_calendar.so' (Asterisk Calendar integration)
    -- Reloading module 'pbx_lua.so' (Lua PBX Switch)
    -- Including switch 'Lua/' in context 'default'
    -- Including switch 'Lua/' in context 'local'
    -- Including switch 'Lua/' in context 'demo'
    -- Including switch 'Lua/' in context 'public'
    -- Including switch 'DUNDi/e164' in context 'ael-dundi-e164-switch'
    -- Time to scan old dialplan and merge leftovers back into the new: 0.010617                                                                                  sec
    -- Time to restore hints and swap in new dialplan: 0.000004 sec
    -- Time to delete the old dialplan: 0.000272 sec
    -- Total time merge_contexts_delete: 0.010893 sec
    -- pbx_lua successfully loaded 37 contexts (enable debug for details).
    -- Reloading module 'res_parking.so' (Call Parking Resource)
    -- Reloading module 'res_config_curl.so' (Realtime Curl configuration)
    -- Reloading module 'res_config_ldap.so' (LDAP realtime interface)
[May 19 23:34:31] NOTICE[22916]: res_config_ldap.c:1832 parse_config: No directo                                                                                 ry user found, anonymous binding as default.
[May 19 23:34:31] ERROR[22916]: res_config_ldap.c:1858 parse_config: No director                                                                                 y URL or host found.
[May 19 23:34:31] NOTICE[22916]: res_config_ldap.c:1776 reload: Cannot reload LD                                                                                 AP RealTime driver.
    -- Reloading module 'res_config_sqlite3.so' (SQLite 3 realtime config engine                                                                                 )
    -- Reloading module 'res_pjsip_authenticator_digest.so' (PJSIP authenticatio                                                                                 n resource)
    -- Reloading module 'res_pjsip_endpoint_identifier_ip.so' (PJSIP IP endpoint                                                                                  identifier)
    -- Reloading module 'res_musiconhold.so' (Music On Hold Resource)
    -- Reloading module 'res_rtp_asterisk.so' (Asterisk RTP Stack)
    -- Reloading module 'res_pjsip_mwi.so' (PJSIP MWI resource)
    -- Reloading module 'res_pjsip_publish_asterisk.so' (PJSIP Asterisk Event PU                                                                                 BLISH Support)
    -- Reloading module 'chan_iax2.so' (Inter Asterisk eXchange (Ver 2))
    -- Reloading module 'chan_mgcp.so' (Media Gateway Control Protocol (MGCP))
 Reloading MGCP
    -- Reloading module 'chan_motif.so' (Motif Jingle Channel Driver)
    -- Reloading module 'chan_ooh323.so' (Objective Systems H323 Channel)
 Reloading H.323
    -- Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP))
 Reloading SIP
    -- Reloading module 'chan_skinny.so' (Skinny Client Control Protocol (Skinny                                                                                 ))
[May 19 23:34:31] NOTICE[22916]: chan_skinny.c:8455 config_load: Configuring ski                                                                                 nny from skinny.conf
    -- Reloading module 'res_adsi.so' (ADSI Resource)
    -- Reloading module 'res_pjsip_notify.so' (CLI/AMI PJSIP NOTIFY Support)
    -- Reloading module 'res_pjsip_outbound_registration.so' (PJSIP Outbound Reg                                                                                 istration Support)
    -- Reloading module 'res_pjsip_phoneprov_provider.so' (PJSIP Phoneprov Provi                                                                                 der)
    -- Reloading module 'app_agent_pool.so' (Call center agent pool applications                                                                                 )
    -- Reloading module 'app_confbridge.so' (Conference Bridge Application)
    -- Reloading module 'app_meetme.so' (MeetMe conference bridge)
    -- Reloading module 'cdr_csv.so' (Comma Separated Values CDR Backend)
    -- Reloading module 'cdr_custom.so' (Customizable Comma Separated Values CDR                                                                                  Backend)
    -- Reloading module 'cdr_manager.so' (Asterisk Manager Interface CDR Backend                                                                                 )
    -- Reloading module 'cel_custom.so' (Customizable Comma Separated Values CEL                                                                                  Backend)
[May 19 23:34:31] NOTICE[22916]: cel_custom.c:97 load_config: No mappings found                                                                                  in cel_custom.conf. Not logging CEL to custom CSVs.
 Added CEL CSV mapping for 0 files.
    -- Reloading module 'cel_manager.so' (Asterisk Manager Interface CEL Backend                                                                                 )
    -- Reloading module 'app_alarmreceiver.so' (Alarm Receiver for Asterisk)
    -- Reloading module 'app_amd.so' (Answering Machine Detection Application)
    -- Reloading module 'codec_speex.so' (Speex Coder/Decoder)
    -- Reloading module 'app_followme.so' (Find-Me/Follow-Me Application)
    -- Reloading module 'app_minivm.so' (Mini VoiceMail (A minimal Voicemail e-m                                                                                 ail System))
    -- Reloading module 'app_osplookup.so' (Open Settlement Protocol Application                                                                                 s)
    -- Reloading module 'app_playback.so' (Sound File Playback Application)
    -- Reloading module 'chan_unistim.so' (UNISTIM Protocol (USTM))
 Reloading unistim.conf...
    -- Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail System))
    -- Reloading module 'res_clialiases.so' (CLI Aliases)
    -- Reloading module 'pbx_ael.so' (Asterisk Extension Language Compiler)
  == Setting global variable 'CONSOLE-AEL' to '"Console/dsp"'
  == Setting global variable 'IAXINFO-AEL' to 'guest'
  == Setting global variable 'OUTBOUND-TRUNK' to '"Zap/g2"'
  == Setting global variable 'OUTBOUND-TRUNKMSD' to '1'
    -- Including switch 'DUNDi/e164' in context 'ael-dundi-e164-switch'
    -- Including switch 'Lua/' in context 'public'
    -- Including switch 'Lua/' in context 'demo'
    -- Including switch 'Lua/' in context 'local'
    -- Including switch 'Lua/' in context 'default'
    -- Time to scan old dialplan and merge leftovers back into the new: 0.001187                                                                                  sec
    -- Time to restore hints and swap in new dialplan: 0.000002 sec
    -- Time to delete the old dialplan: 0.000184 sec
    -- Total time merge_contexts_delete: 0.001373 sec
    -- pbx_ael successfully loaded 37 contexts (enable debug for details).
    -- Reloading module 'pbx_config.so' (Text Extension Configuration)
  == Setting global variable 'CONSOLE' to 'Console/dsp'
  == Setting global variable 'TRUNK' to 'DAHDI/r1'
  == Setting global variable 'TRUNKX' to 'DAHDI/r2'
  == Setting global variable 'TRUNKIAX' to 'IAX2/ASTtest1:test@10.10.10.16:4569'
  == Setting global variable 'TRUNKIAX1' to 'IAX2/ASTtest1:test@10.10.10.16:4569                                                                                 '
  == Setting global variable 'TRUNKBINFONE' to 'IAX2/1112223333:PASSWORD@iax.bin                                                                                 fone.com'
  == Setting global variable 'SIPtrunk' to 'SIP/1234:PASSWORD@sip.provider.net'
  == Setting global variable 'TRUNKloop' to 'IAX2/ASTloop:demo@127.0.0.1:40569'
  == Setting global variable 'TRUNKblind' to 'IAX2/ASTblind:demo@127.0.0.1:41569                                                                                 '
  == Setting global variable 'TRUNKplay' to 'IAX2/ASTplay:demo@127.0.0.1:42569'
  == Setting global variable 'Telnyx' to 'SIP/telnyx'
[May 19 23:34:31] WARNING[22916]: pbx.c:7059 add_priority: Unable to register ex                                                                                 tension '102' priority 1 in 'vicidial-auto-phones', already in use
[May 19 23:34:31] WARNING[22916]: pbx_config.c:1857 pbx_load_config: Unable to r                                                                                 egister extension at line 469 of /etc/asterisk/extensions-vicidial.conf
[May 19 23:34:31] WARNING[22916]: pbx.c:7059 add_priority: Unable to register ex                                                                                 tension '102' priority 2 in 'vicidial-auto-phones', already in use
[May 19 23:34:31] WARNING[22916]: pbx_config.c:1857 pbx_load_config: Unable to r                                                                                 egister extension at line 470 of /etc/asterisk/extensions-vicidial.conf
[May 19 23:34:31] WARNING[22916]: pbx.c:7059 add_priority: Unable to register ex                                                                                 tension '102' priority 3 in 'vicidial-auto-phones', already in use
[May 19 23:34:31] WARNING[22916]: pbx_config.c:1857 pbx_load_config: Unable to r                                                                                 egister extension at line 471 of /etc/asterisk/extensions-vicidial.conf
    -- Including switch 'Lua/' in context 'default'
    -- Including switch 'Lua/' in context 'local'
    -- Including switch 'Lua/' in context 'demo'
    -- Including switch 'Lua/' in context 'public'
    -- Including switch 'DUNDi/e164' in context 'ael-dundi-e164-switch'
    -- Time to scan old dialplan and merge leftovers back into the new: 0.000387                                                                                  sec
    -- Time to restore hints and swap in new dialplan: 0.000003 sec
    -- Time to delete the old dialplan: 0.000161 sec
    -- Total time merge_contexts_delete: 0.000551 sec
    -- pbx_config successfully loaded 37 contexts (enable debug for details).
    -- Reloading module 'pbx_dundi.so' (Distributed Universal Number Discovery (                                                                                 DUNDi))
    -- Reloading module 'res_http_post.so' (HTTP POST support)
    -- Reloading module 'codec_dahdi.so' (Generic DAHDI Transcoder Codec Transla                                                                                 tor)
    -- Reloading module 'app_queue.so' (True Call Queueing)
[May 19 23:34:31] NOTICE[22916]: app_queue.c:8725 reload_queue_rules: queuerules                                                                                 .conf has not changed since it was last loaded. Not taking any action.
Reliably Transmitting (NAT) to 192.76.120.10:5060:
OPTIONS sip:sip.telnyx.com SIP/2.0
Via: SIP/2.0/UDP 172.XXX.11.153:5060;branch=z9hG4bK3d3da707;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.26.11.153>;tag=as791a4d1a
To: <sip:sip.telnyx.com>
Contact: <sip:asterisk@172.26.11.153:5060>
Call-ID: 441d982e4f1719b7544933242b3af920@172.26.11.153:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.32.0-vici
Date: Wed, 20 May 2020 03:34:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                 H, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.76.120.10:5060 --->
SIP/2.0 200 Keepalive
Via: SIP/2.0/UDP 172.XXX.11.153:5060;branch=z9hG4bK3d3da707;rport=5060;received=1                                                                               
From: "asterisk" <sip:asterisk@172.XXX.11.153>;tag=as791a4d1a
To: <sip:sip.telnyx.com>;tag=dfb4940bfc7117e4d7fa62ed6ef36d37.fdce
Call-ID: 441d982e4f1719b7544933242b3af920@172.26.11.153:5060
CSeq: 102 OPTIONS
Server: kamailio (5.0.8 (x86_64/linux))
Content-Length: 0
 

[carpenox]

VERSION: 2.14-753a
BUILD: 200518-0828
SVN Version: 3249
Asterisk 13.32.0


Everythings working great so far: CLI output during call

Code: Select all

Audio is at 15906
Adding codec ulaw to SDP
Adding codec g729 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.76.XXX.10:5060:
INVITE sip:195494XX572@sip.telnyx.com SIP/2.0
Via: SIP/2.0/UDP 172.26.11.153:5060;branch=z9hG4bK29393232;rport
Max-Forwards: 70
From: "V5192347270000005014" <sip:0000000000@172.26.11.153>;tag=as3f11ef2f
To: <sip:19549477572@sip.telnyx.com>
Contact: <sip:0000000000@172.26.11.153:5060>
Call-ID: 0e0b5c0a4a6f97190132b8f560fab72b@172.26.11.153:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.32.0-vici
Date: Wed, 20 May 2020 03:47:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "V5192347270000005014" <sip:0000000000@172.26.11.153>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 306

v=0
o=root 1814195839 1814195839 IN IP4 172.26.11.153
s=Asterisk PBX 13.32.0-vici
c=IN IP4 172.26.11.153
t=0 0
m=audio 15906 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:192.76.XX0.10:5060 --->
SIP/2.0 100 Telnyx trying
Via: SIP/2.0/UDP 172.26.11.153:5060;branch=z9hG4bK29393232;rport=5060;received=15.2XX.125.243
From: "V5192347270000005014" <sip:0000000000@172.26.11.153>;tag=as3f11ef2f
To: <sip:19549XX7572@sip.telnyx.com>
Call-ID: 0e0b5c0a4a6f97190132b8f560fab72b@172.26.11.153:5060
CSeq: 102 INVITE
Server: kamailio (5.0.8 (x86_64/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.76.120.10:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.26.11.153:5060;received=15.2XX.125.243;branch=z9hG4bK29393232;rport=5060
From: "V5192347270000005014" <sip:0000000000@172.26.11.153>;tag=as3f11ef2f
To: <sip:19549XX7572@sip.telnyx.com>;tag=564yyF3HrUg8K
Call-ID: 0e0b5c0a4a6f97190132b8f560fab72b@172.26.11.153:5060
CSeq: 102 INVITE
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
Supported: path, replaces
Allow-Events: talk, hold, conference, refer
Proxy-Authenticate: Digest realm="sip.telnyx.com", nonce="deea206f-ac73-48e9-bc7e-05ff12f4e0bf", algorithm=MD5, qop="auth", opaque="224c64ce-7d7c-4362-91b8-fe98e6c1c08b/10.15.56.4"
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Transmitting (NAT) to 192.76.120.10:5060:
ACK sip:19549477572@sip.telnyx.com SIP/2.0
Via: SIP/2.0/UDP 172.26.11.153:5060;branch=z9hG4bK29393232;rport
Max-Forwards: 70
From: "V5192347270000005014" <sip:0000000000@172.26.11.153>;tag=as3f11ef2f
To: <sip:19549477572@sip.telnyx.com>;tag=564yyF3HrUg8K
Contact: <sip:0000000000@172.26.11.153:5060>
Call-ID: 0e0b5c0a4a6f97190132b8f560fab72b@172.26.11.153:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.32.0-vici
Content-Length: 0


---
[May 19 23:47:28] NOTICE[22763][C-00000000]: chan_sip.c:24170 handle_response_invite: Failed to authenticate on INVITE to '"V5192347270000005014" <sip:0000000000@172.26.11.153>;tag=as3f11ef2f'
Really destroying SIP dialog '0e0b5c0a4a6f97190132b8f560fab72b@172.26.11.153:5060' Method: INVITE
Reliably Transmitting (NAT) to 192.76.120.10:5060:
OPTIONS sip:sip.telnyx.com SIP/2.0
Via: SIP/2.0/UDP 172.26.11.153:5060;branch=z9hG4bK135892be;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.26.11.153>;tag=as47a9441a
To: <sip:sip.telnyx.com>
Contact: <sip:asterisk@172.26.11.153:5060>
Call-ID: 12c358df1a8722e270777acf38d11f6d@172.26.11.153:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.32.0-vici
Date: Wed, 20 May 2020 03:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.76.XX0.10:5060 --->
SIP/2.0 200 Keepalive
Via: SIP/2.0/UDP 172.26.11.153:5060;branch=z9hG4bK135892be;rport=5060;received=15.2XX.125.243
From: "asterisk" <sip:asterisk@172.26.11.153>;tag=as47a9441a
To: <sip:sip.telnyx.com>;tag=dfb4940bfc7117e4d7fa62ed6ef36d37.c60d
Call-ID: 12c358df1a8722e270777acf38d11f6d@172.26.11.153:5060
CSeq: 102 OPTIONS
Server: kamailio (5.0.8 (x86_64/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '12c358df1a8722e270777acf38d11f6d@172.26.11.153:5060' Method: OPTIONS


[carpenox]

phpMyAdmin wasnt loading when i loaded a fresh install of 9.0.2 from .iso - i had to move the directory from /usr/share to /srv/www/htdocs to get it working

asterisk 13.32.0-vici is installed on this new server, svn upgrade to 3252 and db schema to 1596 which ii dont think had anything to do with phpmyadmin not running, but have you checked to see if it works after a fresh install of vicibox?

-Nox

[williamconley]

phpMyAdmin wasnt loading when i loaded a fresh install of 9.0.2 from .iso - i had to move the directory from /usr/share to /srv/www/htdocs to get it working

asterisk 13.32.0-vici is installed on this new server, svn upgrade to 3252 and db schema to 1596 which ii dont think had anything to do with phpmyadmin not running, but have you checked to see if it works after a fresh install of vicibox?

-Nox

Not really related to the topic of "sandbox project" as much as it could be, is it? Sounds like a new topic.

Plus "wasn't loading" is a wee bit vague. Was the apache vhost .conf file for phpmyadmin created? Was it pointing to the wrong folder? Or did it not load for some reason?

[carpenox]

Bill,

Only reason I left that here was because I am running asterisk 13.32.0 from the sandbox project. I am pretty sure its not related to the problem, but either way its fixed, however, the installation for vicibox9.0.2 creates the vhost conf file when u choose to install phpmyadmin or it is supposed to, is it not? The only change was installing 13.32.0

[williamconley]

Thus the suggestion that you create a new thread. This discussion is not likely to be related to the previous. You are there and have the install in question. Did it create the vhost? I certainly wouldn't know. Should it? Moot point (no reason to verify that likelihood) if it DID and the vhost didn't work for some reason. Then we'd be discussing WHY it didn't work. In a thread entirely unrelated to "sandbox project" (which obviously should have had a more descriptive subject line ... at least IMHO)

[Kumba]

phpMyAdmin is such a huge security risk that it's not enabled by default. You will have to manually enable it.

The risk is too great that someone will enable it not knowing or fully understanding how bad phpMyAdmin is security wise, so I made the choice to no longer enable it by default.

So, phpMyAdmin not working isn't a bug, it's a feature

[carpenox]

Kumba,


OK thank you, As far as it nt being enabled by default, isnt that one of the choices IS to enable it during installation? I don't recall exactly if it said install and enable or just to install it... Either way when enabling it, what is the best method to do so? I would imagine moving the folder over to htdocs isnt the best way... lol

-Nox

[carpenox]

Kumba, I updated to 13.34.0 but i noticed the vicibox-install-1 what is that for?

i will have this running during production on a clients server tomorrow so ill let you know if i have any bugs...

Nox

[carpenox]

Kumba,

I did the update from the 3rd and since then my meetme conferences are not working. any ideas? I have tried to get this fixed as well and I have had no luck. Any suggestions to get the conferences working again would be appreciated.

Nox

[williamconley]

Kumba,

I did the update from the 3rd and since then my meetme conferences are not working. any ideas? I have tried to get this fixed as well and I have had no luck. Any suggestions to get the conferences working again would be appreciated.

Nox

Not working is a wee bit vague. Is there an error message? Or are the meetme.conf entries created ...? Does agent login fail? (and if so ... the error message in the cli ...)

[Kumba]

Check to see if dahdi is loaded.

[carpenox]

its working again, i forgot to update this post, i forgot what i did to fix it

[carpenox]

Kumba, What changed in the vicibox-dynportal that was just updated? Just curious if I should run it on my existing servers or not. Thx

Nox

[Kumba]

Kumba, What changed in the vicibox-dynportal that was just updated? Just curious if I should run it on my existing servers or not. Thx

Mostly some apache conf and spec file changes to make it more maintainable as a package. You can download the config file and tarball and run a diff over them or look at the commit changes in OBS to see what was changed. You will likely need to make a login on OBS to view commits though. Downloading the sources should still be publicly available.

[carpenox]

ok cool thx

[carpenox]

Kumba, gm sir, i seen u switched it to 13.38.0, anything i should know? i already moved one server into production and the only thing i noticed was this:

[Nov 29 08:13:37] WARNING[29571]: chan_iax2.c:1217 iax_error_output: Information element length exceeds message size
[Nov 29 08:13:37] WARNING[29571]: chan_iax2.c:10210 socket_process_helper: Undecodable frame received from 'server ip'

[carpenox]

kumba, the last update broke something, ast manager isnt working right anymore since asterisk-13.38.0-lp151.34.8.x86_64