I have a older roll my own install debian 7 freepbx 12 asterisk 11
its been running great for what we need I upgraded the asterisk to
11.6-cert18

and also tried to fix the framework in the freepbx module admin that the dashboard wanted me to upgrade.

so i did this and it would not apply
I tried disabling things mentioned in the log files to see if I could get past apply
[2018-Jul-14 19:38:35] [PHP-NOTICE] (/var/lib/asterisk/bin/retrieve_conf:181) - Use of undefined constant FREEPBX_LOG_ERROR - assumed ‘FREEPBX_LOG_ERROR’
[2018-Jul-14 19:38:35] [FREEPBX_LOG_ERROR] (bin/retrieve_conf:181) - Tried to link /var/www/html/admin/modules/endpointman/ari/modules/phonesettings.module to /var/www/html/recordings/modules/phonesettings.module, but /var/www/html/recordings/modules doesn’t exist
[2018-Jul-14 19:38:35] [FATAL] (libraries/utility.functions.php:470) - retreive_conf failed to get engine information and cannot configure up a softwitch with out it. Error: ERROR-UNABLE-TO-PARSE
[2018-Jul-14 19:38:35] [CRITICAL] (BMO/Notifications.class.php:493) - [NOTIFICATION]-[freepbx]-[RCONFFAIL] - retrieve_conf failed, config not applied (Reload failed because retrieve_conf encountered an error: 1)
[2018-Jul-14 19:41:29] [PHP-NOTICE] (/var/www/html/admin/libraries/modulefunctions.class.php:1031) - Undefined offset: 4
[2018-Jul-14 19:41:29] [PHP-NOTICE] (/var/www/html/admin/libraries/modulefunctions.class.php:1031) - Undefined offset: 4
[2018-Jul-14 19:41:54] [PHP-NOTICE] (/var/www/html/admin/page.modules.php:191) - ob_flush(): failed to flush buffer. No buffer to flush
[2018-Jul-14 19:41:54] [PHP-NOTICE] (/var/www/html/admin/page.modules.php:367) - ob_flush(): failed to flush buffer. No buffer to flush
[2018-Jul-14 19:41:54] [PHP-NOTICE] (/var/www/html/admin/config.php:544) - ob_end_flush(): failed to delete and flush buffer. No buffer to delete or flush
[2018-Jul-14 19:41:56] [PHP-NOTICE] (/var/www/html/admin/libraries/modulefunctions.class.php:1031) - Undefined offset: 4
[2018-Jul-14 19:41:56] [PHP-NOTICE] (/var/www/html/admin/libraries/modulefunctions.class.php:1031) - Undefined offset: 4
[2018-Jul-14 19:41:59] [NOTICE] (BMO/Notifications.class.php:493) - [NOTIFICATION]-[core]-[MEMLIMIT] - Memory Limit Changed (Your memory_limit, -1M, is set too low and has been increased to 100M. You may want to change this in you php.ini config file)
[2018-Jul-14 19:41:59] [FATAL] (libraries/utility.functions.php:470) - retreive_conf failed to get engine information and cannot configure up a softwitch with out it. Error: ERROR-UNABLE-TO-PARSE
[2018-Jul-14 19:41:59] [CRITICAL] (BMO/Notifications.class.php:493) - [NOTIFICATION]-[freepbx]-[RCONFFAIL] - retrieve_conf failed, config not applied (Reload failed because retrieve_conf encountered an error: 1)
[2018-Jul-14 19:42:01] [NOTICE] (BMO/Notifications.class.php:493) - [NOTIFICATION]-[core]-[MEMLIMIT] - Memory Limit Changed (Your memory_limit, -1M, is set too low and has been increased to 100M. You may want to change this in you php.ini config file)
[2018-Jul-14 19:42:01] [FATAL] (libraries/utility.functions.php:470) - retreive_conf failed to get engine information and cannot configure up a softwitch with out it. Error: ERROR-UNABLE-TO-PARSE
[2018-Jul-14 19:42:01] [CRITICAL] (BMO/Notifications.class.php:493) - [NOTIFICATION]-[freepbx]-[RCONFFAIL] - retrieve_conf failed, config not applied (Reload failed because retrieve_conf encountered an error: 1)

I am looking for a simple fix.

i backed up all the files and tried to us the command I used to install it according to my notes
./install_amp --installdb --username=asteriskuser --password=S0MePassworDDD --dbhost=‘127.0.0.1’ --dbname=‘asterisk’ --freepbxip=‘192.168.1.210’ --cgibin=’/var/www/cgi-bin’ --bin=’/var/lib/asterisk/bin’ --sbin=’/usr/local/sbin’ --asteriskuser=admin --asteriskpass=B8nKmYaSTdbP855 --asteriskip=‘127.0.0.1’ --scripted

everything seems fine until it dies with this
Checking for /etc/asterisk/asterisk.conf…OK
Reading /etc/asterisk/asterisk.conf…OK
Using asterisk as PBX Engine
Checking for Asterisk version…[FATAL] Could not determine asterisk version (got: “Asterisk certified/11.6-cert18” please report this)

help

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

Forget about generating certificate, Let’s Encrypt now supports wildcard SSLs the caveat is you have to update your DNS record manually every 90 days Follow the instructions on sslforfree[dot]com (this is a site approved by LE, please check “SSL for free” on LE website. I can’t link to it because I’m new user but you can find the page by searching for “acme client implementations” on google.

I’m happy to help on further questions, good luck

You need to upgrade to a supported FreePBX version. Your FreePBX 12 is not supported and it doesn’t support newer asterisk versions like what you are trying to use.

I just want it to run can I not just force it some how? editing something or a option i tried --force-version and it seems to be a real option but just ignors it

–force-version=‘11.6-cert18’
–force-version=‘Asterisk certified/11.6-cert18’

I am trying not to reload the whole thing. all I did was update a module because the dashboard made it sound important worst thing I could of done.

now I can even change a ext or anything because I am unable to apply anything

I am running wheezy Debian 7 I have a feeling freepbx requires something that might need me to upgrade to debian 8 i do not want to do that. The reason i think this is i am not able to find any mention of freepbx 13 and debian 7 in the same sentence in any searches i have done

I will attempt to upgrade it right now if I am confident that I can do so.

don’t forget to make your thread resolved if it is.

so far I attempted to install what is nessary and download the compressed 13 latest

but when I try to install

Checking if Asterisk is running and we can talk to it as the ‘asterisk’ user…Error!
Could not determine Asterisk version (got: Asterisk certified/11.6-cert18 built by root @ pb1 on a x86_64 running Linux on 2018-07-14 15:01:51 UTC). Please report this.
Done
Preliminary checks done. Starting FreePBX Installation
Checking if this is a new install…Yes (No /etc/freepbx.conf file detected)
Database installation checking credentials and permissions…Connected!
Initializing FreePBX Settings
Finished initalizing settings
Copying files (this may take a bit)…
0/5291 [>---------------------------] 0%/etc/asterisk/voicemail.conf.template has been changed from the original version.
Overwrite:
[x] Exit
[y] Yes
[n] No
[d] Diff

y
-> Original file: /etc/asterisk/voicemail.conf.template
-> New file: /usr/src/freepbx/amp_conf/astetc/voicemail.conf.template
Exiting install program.

Hi I am getting this error when trying to “Apply Config” after an upgrade…

exit: 1
Unable to continue. mkdir(): Permission denied in /var/www/html/admin/modules/music/Music.class.php on line 161
#0 [internal function]: Whoops\Run->handleError(2, ‘mkdir(): Permis…’, ‘/var/www/html/a…’, 161, Array)
#1 /var/www/html/admin/modules/music/Music.class.php(161): mkdir(’/var/lib/asteri…’, 511, true)
#2 /var/www/html/admin/libraries/BMO/FileHooks.class.php(97): FreePBX\modules\Music->genConfig()
#3 /var/www/html/admin/libraries/BMO/FileHooks.class.php(26): FreePBX\FileHooks->processNewHooks()
#4 /var/lib/asterisk/bin/retrieve_conf(877): FreePBX\FileHooks->processFileHooks(Array)
#5 {main}

sorted

I ran

fwconsole chown

I would do the following:

-Update and reboot your firewall.
-Reboot PBX.
-Send a call log to Vitality, they should be able to tell you whats up.

The only thing in this list that should be done is contacting Vitelity. Calls to a single destination (number) over one particular carrier generally do not mean a firewall or PBX issue.

The real question here: Is this just one number or is this ALL Verizon numbers being called via Vitelity?

Based on the information provided so far, yes this seems to be a Vitelity issue. Randomly rebooting firewalls and the PBX, just because, is not a viable troubleshooting method. Reach out to Vitelity and give them the call example and timestamp of the call, they’ll be able to look at it.

Also keep in mind though, Vitelity is an aggregate and generally doesn’t touch the media (they pass it through to their upstream carrier) so they may or may not be able to give you an answer right away.

I don’t understand why this is requiring the use of custom.conf files or a lot of confusion on how this should be done.

SIPGate is requiring that the CallerID be presented in PAI format and that only SIPGate numbers (i.e. ones on your account) can be presented as CallerID. In other words, they don’t just allow any ole number to be passed as CallerID.

So in this case you would set up at least ONE Outbound Route that has the Route CallerID set to the SIPGate number you want to present as CallerID and then you go into your PJSIP trunk configuration and do this:

2018-07-15 09_35_40-FreePBX Administration.png1234x131 8.28 KB

That will take the Route CallerID from the Outbound Route (or even the Outbound CallerID from the extension if the route allows it) and present it in P-Asserted-Identity format.

Now if this is Chan_SIP trunk then everything is the same except for the trunk settings. In that case you just add (or modify) the sendrpid= setting to be sendrpid=pai and that will present P-Asserted-Identity format for the call.

OP states Sipgate wants P-Preferred-Identity not P-Asserted-Identity.

Oops. My bad.

ok after alot of rm -rf and drop databases and nemerious versions of pjproject (pjsip 2.7 seems to let me compile asterisk 13…)

i get stuck here
I have tried with the root password and asteriskuser criedentials and they work just not in this freepbx wtf?

Database installation checking credentials and permissions…Connected!

[Exception]
SQLSTATE[42000] [1049] Unknown database ‘asterisk’::SQLSTATE[42000] [1049] Unknown database ‘asterisk’

[PDOException]
SQLSTATE[42000] [1049] Unknown database ‘asterisk’

Thank you both for the replies. Yes it seems all calls to a Verizon cell mailbox just hangs. I will open a trouble ticket with them

got by this with
./install --rootdb

Just a guess, your PBX is offering T.38 fax, the beep sounds like a fax tone and Vitelity sends a re-invite to switch to T.38. You can verify this easily with a SIP trace. At the Asterisk command prompt, type
pjsip set logger on
or
sip set debug on
according to your trunk type.
Then make a failing call and the SIP will appear in the Asterisk log.

I have an issue where inbound calls from my trunk work only sometimes. There is no pattern I can discern as to what may be causing this failure. It will work for one call, only to fail the next, just to work on the third one; other times, it will work without error for several hours.

When a call fails, I see “No matching peer” in my logs, followed by returning 401 Unauthorized to my trunk provider.

My provider is Voyant, and I am using IP authentication.

Here is a log snippet of when it works:

<--- SIP read from UDP:137.192.80.33:5060 --->
INVITE sip:+1231597XXXX@206.189.X.X:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 137.192.80.33:5060;branch=z9hG4bKqo2r1n206o9mpatbljp0.1
From: <sip:+1231818XXXX@137.192.80.33:5060;isup-oli=62>;tag=gK0c2c5cff
To: <sip:+1231597XXXX@137.192.80.33:5060>
Call-ID: 1628223420_113246044@199.199.12.56
CSeq: 443724 INVITE
Max-Forwards: 67
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+1231818XXXX@137.192.80.33:5060;transport=udp>
P-Asserted-Identity: "CNAM " <sip:+1231818XXXX@137.192.80.33:5060>
Supported: timer,100rel,precondition
Session-Expires: 1800
Min-SE: 90
Content-Length: 364
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 77861 169058 IN IP4 137.192.80.33
s=SIP Media Capabilities
c=IN IP4 137.192.80.33
t=0 0
m=audio 25498 RTP/AVP 96 0 18 101 100
a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/48000
a=fmtp:101 0-15
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=maxptime:20
a=record:on
<------------->
[2018-07-15 15:25:18] VERBOSE[17369] chan_sip.c: --- (17 headers 16 lines) ---
[2018-07-15 15:25:18] VERBOSE[17369] chan_sip.c: Sending to 137.192.80.33:5060 (no NAT)
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Sending to 137.192.80.33:5060 (no NAT)
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Using INVITE request as basis request - 1628223420_113246044@199.199.12.56
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found peer 'voyant' for '+1231818XXXX' from 137.192.80.33:5060
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] netsock2.c: Using SIP RTP TOS bits 184
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] netsock2.c: Using SIP RTP CoS mark 5
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found RTP audio format 96
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found RTP audio format 0
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found RTP audio format 18
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found RTP audio format 101
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found RTP audio format 100
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found audio description format opus for ID 96
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found audio description format PCMU for ID 0
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found audio description format G729 for ID 18
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found unknown media description format telephone-event for ID 101
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Found audio description format telephone-event for ID 100
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|g726|g722|g723|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|vp9|red|t140|silk|silk|silk|silk), peer - audio=(ulaw|g729|opus)/video=(nothing)/text=(nothing), combined - (ulaw|g729|opus)
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Peer audio RTP is at port 137.192.80.33:25498
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: Looking for +1231597XXXX in from-pstn-e164-us (domain 206.189.X.X)
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] sip/route.c: sip_route_dump: route/path hop: <sip:+1231818XXXX@137.192.80.33:5060;transport=udp>
[2018-07-15 15:25:18] VERBOSE[17369][C-00000334] chan_sip.c: 
<--- Transmitting (no NAT) to 137.192.80.33:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 137.192.80.33:5060;branch=z9hG4bKqo2r1n206o9mpatbljp0.1;received=137.192.80.33
From: <sip:+1231818XXXX@137.192.80.33:5060;isup-oli=62>;tag=gK0c2c5cff
To: <sip:+1231597XXXX@137.192.80.33:5060>
Call-ID: 1628223420_113246044@199.199.12.56
CSeq: 443724 INVITE
Server: FPBX-13.0.195.4(13.21.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+1231597XXXX@206.189.X.X:5060>
Content-Length: 0

And when it doesn’t:

<--- SIP read from UDP:137.192.80.33:5060 --->
INVITE sip:+1231597XXXX@206.189.X.X:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 137.192.80.33:5060;branch=z9hG4bKostdol30boa06ev7lud0.1
From: <sip:+1231818XXXX@137.192.80.33:5060;isup-oli=62>;tag=gK0c3466a1
To: <sip:+1231597XXXX@137.192.80.33:5060>
Call-ID: 1628224886_117117091@199.199.12.56
CSeq: 529262 INVITE
Max-Forwards: 67
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+1231818XXXX@137.192.80.33:5060;transport=udp>
P-Asserted-Identity: "CNAM " <sip:+1231818XXXX@137.192.80.33:5060>
Supported: timer,100rel,precondition
Session-Expires: 1800
Min-SE: 90
Content-Length: 365
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 600124 360022 IN IP4 137.192.80.33
s=SIP Media Capabilities
c=IN IP4 137.192.80.33
t=0 0
m=audio 25440 RTP/AVP 96 0 18 101 100
a=rtpmap:96 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/48000
a=fmtp:101 0-15
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=maxptime:20
a=record:on
<------------->
[2018-07-15 15:31:43] VERBOSE[17369] chan_sip.c: --- (17 headers 16 lines) ---
[2018-07-15 15:31:43] VERBOSE[17369] chan_sip.c: Sending to 137.192.80.33:5060 (no NAT)
[2018-07-15 15:31:43] VERBOSE[17369][C-00000339] chan_sip.c: Sending to 137.192.80.33:5060 (no NAT)
[2018-07-15 15:31:43] VERBOSE[17369][C-00000339] chan_sip.c: Using INVITE request as basis request - 1628224886_117117091@199.199.12.56
[2018-07-15 15:31:43] VERBOSE[17369][C-00000339] chan_sip.c: No matching peer for '+1231818XXXX' from '137.192.80.33:5060'
[2018-07-15 15:31:43] VERBOSE[17369][C-00000339] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to 137.192.80.33:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 137.192.80.33:5060;branch=z9hG4bKostdol30boa06ev7lud0.1;received=137.192.80.33
From: <sip:+1231818XXXX@137.192.80.33:5060;isup-oli=62>;tag=gK0c3466a1
To: <sip:+1231597XXXX@137.192.80.33:5060>;tag=as57460408
Call-ID: 1628224886_117117091@199.199.12.56
CSeq: 529262 INVITE
Server: FPBX-13.0.195.4(13.21.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="44b2217b"
Content-Length: 0

PEER Details:

type=peer
trustrpid=no
sendrpid=no
insecure=port,invite
host=X.st.sip.global
dtmfmode=rfc2833
;disallow=all
context=from-pstn-e164-us
canreinvite=no
;allow=ulaw
allow=all

I changed the allow/disallow settings thinking that the trunk may be choosing random codecs (grasping at straws), but both configurations work the same.

I have read through other posts reporting similar issues, but those seemed to happen when the provider is using more than one IP address for invites, but that isn’t the case with me.

It may be important to know that I also have a Vitelity trunk configured, which is working fine.

I would appreciate any input on this issue.