@slondono wrote:

Hello,

I have problems with the way the call is received by one of the servers, which runs asterisk 11.0.
Asterisk is receiving the call from the from-trunk context, Ignoring the from-internal context specified in the configuration of the incoming peer
The context parameter has no effect, I need calls to come in with from-internal To be able to made calls through the outgoing routes of this server from server2.

¿Any idea to force asterisk to use from-internal context for this trunk?

Best regards,

Santiago.

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@cynjut wrote:

Quick one.

Just got a call from an old customer - they want to upgrade from the PIAF system I installed for them in 2012. Asterisk 1.8.7.2, Core version 2.10.0.0.

Short of a forklift upgrade, what's the webpage link that has the upgrade scripts to bring this bad boy up from where I left it. I know I'm going to have to download at least the first upgrade module in the list, since the system can't get to the FreePBX module repository.

For those keeping score, they brought me in to fix an Asterisk 1.2 system that a previous IT guy had built and now they want me to do a second round of upgrades.... This is going to be fun.

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@Bullfrog wrote:

I am currently running FreePBX Distro 10.13.66-18. I also use the System Admin Pro Module. Within the module, I have enabled Automatic Updates to keep my system reasonably current with up-to-date patches and security fixes. However, for the last six months or so, every time an automatic update is performed, it always results in the update getting "stuck". When checking the log, I see this error:

Update currently in progress! Please wait until complete.

2017-02-05 05:00:12: (0) Update already in progress (locked)
2017-02-15 07:15:41: (10.13.66-18) Requesting updates
2017-02-15 07:15:42: (10.13.66-18) Processing update
2017-02-15 07:15:42: (10.13.66-18) Now running 10.13.66-18 script
2017-02-15 07:15:42: (10.13.66-18) Starting stage 1
2017-02-15 07:22:03: (10.13.66-18) Stage 1 complete
2017-02-15 07:22:03: (10.13.66-18) Starting stage 2
2017-02-15 07:24:41: (10.13.66-18) Stage 2 complete
2017-02-15 07:24:41: (10.13.66-18) Stage 3 complete
2017-02-15 07:24:41: (10.13.66-18) Starting stage 4
2017-02-15 07:24:43: (10.13.66-18) Stage 4 complete
2017-02-15 07:24:44: (10.13.66-18) Starting stage 5
2017-02-15 07:24:44: (10.13.66-18) Stage 5 complete
2017-02-15 07:24:44: (10.13.66-18) Starting stage 6
2017-02-15 07:24:44: (10.13.66-18) Stage 6 complete
2017-02-15 07:24:44: (10.13.66-18) Upgrade script complete
2017-02-15 07:24:44: (10.13.66-18) Upgrade process successfully completed!
2017-02-26 05:00:15: (0) Update already in progress (locked)
2017-03-05 05:00:04: (10.13.66-19) Update already in progress (locked)
2017-03-12 05:00:09: (10.13.66-19) Update already in progress (locked)

You can see in that log above where the update to 10.13.66-18 was stalled. I unlocked it, and was able to get it to finish, but as soon as 10.13.66-19 was release, the problem just repeats. The button in the System Admin module that allows me to perform the update immediately is grayed out with the message, "Upgrade in progress".

I know I can correct the issue according to Lorne's suggestion:

Does anyone know what may be causing this to happen with every upgrade? Is it unique to my install? Or are others also experiencing a similar recurrence of the locked or stuck updates?

I would love to troubleshoot this and get it resolved. I would appreciate any suggestions you have.

--

Jeremy

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@FreeSWServers wrote:

Hello,

I have setup FreePBX 14 on SNG7 official Distro, and I am getting the following error when I watch a call in the asterisk CLI.

[2017-03-17 14:09:10] WARNING[16629]: res_odbc.c:460 ast_odbc_print_errors: SQL Execute returned an error: 23000 [MySQL][ODBC 5.2(w) Driver][mysqld-5.5.52-MariaDB]Duplicate entry '0' for key 'PRIMARY'
[2017-03-17 14:09:10] WARNING[16629]: res_odbc.c:404 ast_odbc_prepare_and_execute: SQL Execute error -1!
[2017-03-17 14:09:10] WARNING[16629]: cel_odbc.c:773 odbc_log: Insert failed on 'asteriskcdrdb:cel'.  CEL failed: INSERT INTO cel

Any idea what is going on here>? I'm not sure how to debug this..

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@voipmuch wrote:

Hello FreePBX fans,

We have everything running...
But need to query if an IP is blocked (from command line).

There appears to not be any zone for the blocked attackers... or at least I cannot figure out what it is.

fwconsole firewall list blacklist only works to show blacklisted IP (which are manually added).
fwconsole firewall list blocked does not work, as blocked is not a valid zone.

1) How do we list the blocked attackers from command line?

2) Depending on the answer to #1, to add and remove an IP from command line, you need the zone.

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@igorek600 wrote:

Hello, I have successfully installed Freepbx 13 On CentOS 6.8 x64 on Digital Ocean but have 1 issue. After restart I get reload config error, to fix it I need to run manual command:
asterisk -rx 'core stop now'
fwconsole start

Once it runs almost everything works perfectly (except firewall issues and system admin won't show version I have and won't update it)

Is there a way to setup this command to run on boot?

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@v4704 wrote:

Asterisk Version: 13.14.0
FreePBX 13.0.190.19
System Firewall: 13.0.44.4

My desktop PC is almost never changing its public IP, so the "trust zone" is doing a good job allowing the connection. My problem is with my cell phone that gets new IP every few days from my cellular provider.

I'm using an Android device with Media5 free client, there is no way the device is sending any type of request with wrong password (I'm not trying to connect to the web interface, SSH or anything else other than SIP from it), but it still going to blocked list.

What can cause that? what are the parameters that gets an IP blocked by the Responsive Firewall?

Many thanks!

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@dickson wrote:

I work with an organization that has 4 different FreePBX systems (divided by different lines of business).
Everything is fine, but voicemail is handled by each server individually, and I would like to know if anyone has ever centralized the voicemail of multiple FreePBX instances into a separate server?

I'll add that I know how i could do this through some massive dialplan hacking, just wondering if there is something in the GUI that i could use to help me trick the diaplan to make this a little easier

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@tonyg wrote:

hi, i am testing a freepbx install and need some help with upgrade procedures. i have learned that there seems to be 3 categories of upgrade?
modules, freepbx and os.

i understand that modules are upgraded in the module admin
i also learned that freepbx upgrade via the freepbx upgrade scripts.

my questions is about the os, what is the recommended way to keep the os up to date? yum? some other way?

thanks in advance!

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@albertxyc wrote:

The problem:
FreePBX 13.0.190.19 Asterisk 11.19.0
I try to do this:
Add inbound route and set moh for example Callcenter-moh. And this inbound route destination set to queue "Callcenter".
In the queue "Callcenter" set moh to "WaitingMusic-moh".
When i try to call to inbound route "Callcenter-moh" doesn't play, but queues moh "waitingmusic-moh" starts play immediately.
If i try to set the inbound route to extension, inbound route moh also doesn't play... only rings
Why he skips moh on inbound route?

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@tonyg wrote:

hi, i have a new install of freepbx 13 (patched) with 15 grandstream 2130 phones. I am using siptls and srtp . the phones are in their own vlan and the pbx is local to them. each phone has 2 accounts (2 extensions). what I am seeing is at random points during the day, one or the other account on the phone will change to unavailable. the account that survives seems to work fine but not the unavailable one (obviously). if i reboot the phone, they are both available again for a while. can anyone give me some hints as to where to start looking to figure out why this is happening. when an extension goes out, this is what comes up on asterisk

-- Contact 7323/sips:7323@172.30.2.52:44924;transport=TLS is now Unreachable.  RTT: 0.000 msec

== Endpoint 7323 is now Unreachable

here is a pjsip show contacts

uepbx1*CLI> pjsip show contacts

Contact:

Contact: 5002/sips:5002@172.30.2.57:40802;transport=TLS 6094490057 Unavail 0.000
Contact: 7002/sips:7002@172.30.2.57:41122;transport=TLS bf147876e9 Avail 11.844

as you can see, one of the extensions on the phone is available and one is not

thanks in advance

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@gforceco wrote:

Hello,

I know this can be done by using the certbot-auto module. However, I don't want to mess with anything and would like to use the module that came with FreePBX.

Is there a way to have a Let's Encrypt certificate be validated for multiple domains through the Certificate Manager module? The reason for this is that I have three clients under the same server. So each client has its own domain pointing to the server. I was able to add a certificate for each domain, but of course, I am only able to select one at a time on HTTPS Setup (SysAdmin Module). Any help will be greatly appreciated. Thanks in advance!

FreePBX Version: 13.0.190.19
Asterisk Version: 11.25.1

Using FreePBX Distro on the cloud, hosted by partner, FreePBXHosting.com

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@carlosf wrote:

Hello,
I'm new to this forum, so if I posted this topic in the wrong place, please forgive me.

I've ran into a problem with my fresh installation of FREEPBX 13.0.191.5.
I just finished installing and when I try to click in "Apply config" the following error is shown:

"Error: Did not receive valid response from server

XHR response code: 500 XHR responseText: undefined jQuery status: error"

My current linux version is Centos 6.8 and the current asterisk version(which I've installed before installing freepbx) is "Asterisk certified/13.13-cert2"

When I try to execute the command "amportal a r" the following error also appears:

[FATAL] retreive_conf failed to get engine information and cannot configure up a softwitch with out it. Error: ERROR-UNABLE-TO-PARSE

Can you guys help me out?

P.S: I just noticed how strange this version name is. But it's what is shown when I type "core show version" in my Asterisk CLI.

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@tonyg wrote:

hi, i have a new install of freepbx 13 (patched) and about 15 grandstream phones. the phones connect using sip-tls and srtp. the phones seem to be working good, but i have noticed that if I make a change (any change) on the pbx that requires me to hit the 'apply config' button, all the phones go into a "unavailable state" according to "pjsip show contacts". at this point, the phones seem fine but the pbx has dropped them and calls go directly to voicemail. the only way I can recover is to reboot all the phones.

this seems like a bug and I would be happy to open a ticket, I just wanted to see if anyone has experienced this and maybe have a simple solution?

thanks

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@pcguru wrote:

I was wondering if anyone out there has successfully accomplished using FreePBX as a SIP router. I have several independent offices, all with their own FreePBX server and unique extension ranges. Now they have joined together under a universal management group and want to be able to dial extensions in other offices directly. I can set up IAX2 trunks and routes between the servers, but it will take awhile as there are over 30 offices. Using IAX2 trunks to siprouter.mycompany.com sends the call to the FreePBX distro acting as a router, but it does not pass the call on to where it needs to go.

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@4allbusiness wrote:

Hello FreePBX and Users,

We are seeing very short audio dropouts at every whole minute and whole hour on systems where we haven 25 or more VM FreePBX-Distro's. Version, memory or other changes like x32 or x64, does not show any difference.
For our feeling, older upgraded Distro's from 5.12.x show this problem lesser than the newest 6.13.x or 10.13.x versions.

After investigating for day's we can conclude this is coming from within the FreePBX vm's

Searching for cronjobs every minute and hour, but not sure what job, or combination, can create this issue.

The network latency shows spikes at the whole minute, probably cause by cronjobs, which are all executed at the exact same time.

Now we looked in the file '/var/www/html/admin/modules/dashboard/scheduler.php'
Which contains the next rule:

// Sleep to fix crazy issues with large VM hosting providers
sleep(mt_rand(1,30));

Does anybody know which scheduled job takes the most time or system resources which can cause this latency problems?

Any help is welcome. Many thanks in advance.

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@tonyg wrote:

hi, i have a new install of 10.13.66.19, with some grandstream phones. the voicemail light is not lighting. I have been searching a lot and stumbled across this:

which indicates that the mailbox has to agree with the section heading in voicemail.conf.

so when I look at my pjsip.aor.conf file, i see my extensions like this:

[5028]
type=aor
mailboxes=5028@device
max_contacts=1
remove_existing=yes
maximum_expiration=7200
minimum_expiration=60
qualify_frequency=60

but when i look at my voicemail.conf, the section looks like this:

[default]
5028 => 1111,person,person@place.com,,attach=no|saycid=no|envelope=no|delete=no

note that voicemail section header is default, while the mailbox is @device.....so, am I right in thinking this is a problem...if so, why is it like this? I do not edit these files, I only make changes in the gui?

thanks for your help

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@jerryriggin wrote:

Hi!

I have a 13.0.190.19 distro with backup module 13.0.26.1. with many custom files in /var/lib/asterisk/agi-bin. They are all owned by asterisk:asterisk and mode 775.

I want to backup freePBX and restore to hot standby via automation. The problem is that I can't seem to configure a backup that includes my custom AGI files.

I have this line in the backup config:
__ASTVARLIBDIR__/agi-bin

I have tried all versions of the config line I can imagine, * in parameters, with and without trailing /, but my custom agi files are never included in the backup.

I created a workaround and tgz the agi-bin separately in my automation scripts, but would like to know what the syntax must be or what else must change to include these files in the backup.

Thanks!

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@tonyg wrote:

Hi, i just put up a FreePBX 14.0.1rc1.8 i have set up ldap user authentication to my active directory and that has imported my users (everything looks ok there). i went to my userid and granted myself admin rights. i then logout and login to admin panel using my ldap creds....this happens: I put this in as a bug, they the kicked it out saying it was a support issue....

it seems to imply that broadcast module is a requirement? is that true, broadcast is a commercial module.....any help would be greatly appreciated.!!!

Exception thrown with message "Unable to locate the FreePBX BMO Class 'Broadcast'A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma install broadcast 2) fwconsole ma enable broadcast"

Stacktrace:

8 Exception in /var/www/html/admin/libraries/BMO/Self_Helper.class.php:216

7 FreePBX\Self_Helper:loadObject in /var/www/html/admin/libraries/BMO/Self_Helper.class.php:104

6 FreePBX\Self_Helper:autoLoad in /var/www/html/admin/libraries/BMO/Self_Helper.class.php:37

5 FreePBX\Self_Helper:__get in /var/www/html/admin/libraries/BMO/FreePBX.class.php:103

4 FreePBX:__callStatic in /var/www/html/admin/modules/broadcast/functions.inc.php:8

3 FreePBX:Broadcast in /var/www/html/admin/modules/broadcast/functions.inc.php:8

2 require_once in /var/www/html/admin/bootstrap.php:372

1 require_once in /etc/freepbx.conf:9

0 include_once in /var/www/html/admin/config.php:100

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@FreeSWServers wrote:

Hello,

I have a very weird issue, nightly I restore a TGZ from Server A to Server B for backup purposes. This seems to be corrupting the Postfix main.cf file. I have tracked it down to happening immediatley after restoring the TGZ, which I restore via the CLI via the following:

/usr/bin/php /var/lib/asterisk/bin/restore.php --restore=/data/backups/fpbx/voip1.tgz --items=all

I posted a Question on Unix Linux, but now I realized its more of a FreePBX thing.

Here is the main.cf and how I installed it:

cat << 'EOL' >/etc/postfix/main.cf
#
# PostFix Config

##Modify Below [HOSTNAME]
myhostname = [HOSTNAME]

smtpd_banner = $myhostname ESMTP $mail_name
biff = no
append_dot_mydomain = no
readme_directory = no
smtpd_tls_cert_file=/etc/ssl/certs/ssl-cert-snakeoil.pem
smtpd_tls_key_file=/etc/ssl/private/ssl-cert-snakeoil.key
smtpd_use_tls=yes
smtpd_relay_restrictions = permit_mynetworks permit_sasl_authenticated defer_unauth_destination
smtpd_tls_session_cache_database = btree:${data_directory}/smtpd_scache
smtp_tls_session_cache_database = btree:${data_directory}/smtp_scache
smtp_tls_CAfile = /etc/ssl/certs/ca-certificates.crt
smtp_use_tls=yes
smtp_sasl_auth_enable = yes
smtp_sasl_security_options = noanonymous
smtp_sasl_password_maps = hash:/etc/postfix/sasl/sasl_passwd
mynetworks = 127.0.0.0/8 [::ffff:127.0.0.0]/104 [::1]/128
mailbox_size_limit = 0
recipient_delimiter = +
inet_interfaces = all
inet_protocols = all
alias_maps = hash:/etc/aliases
alias_database = hash:/etc/aliases
myorigin = /etc/mailname
canonical_maps = hash:/etc/postfix/canonical
relayhost = [smtp.hmc1.comcast.net]:587
mydomain = domain.com
mydestination = $myhostname $myhostname.$mydomain localhost.$mydomain localhost
sender_canonical_maps = static:$myhostname@$mydomain
EOL

Here is what my main.cf looks like after it gets randomly messed up, this is consistently how it looks as well:

#
# PostFix Config

##Modify Below [HOSTNAME]


smtpd_banner = $myhostname ESMTP $mail_name
biff = no
append_dot_mydomain = no
readme_directory = no
smtpd_tls_cert_file=/etc/ssl/certs/ssl-cert-snakeoil.pem
smtpd_tls_key_file=/etc/ssl/private/ssl-cert-snakeoil.key
smtpd_use_tls=yes
smtpd_relay_restrictions = permit_mynetworks permit_sasl_authenticated defer_unauth_destination
smtpd_tls_session_cache_database = btree:${data_directory}/smtpd_scache
smtp_tls_session_cache_database = btree:${data_directory}/smtp_scache





mynetworks = 127.0.0.0/8 [::ffff:127.0.0.0]/104 [::1]/128
mailbox_size_limit = 0
recipient_delimiter = +
inet_interfaces = all
inet_protocols = all
alias_maps = hash:/etc/aliases
alias_database = hash:/etc/aliases

canonical_maps = hash:/etc/postfix/canonical


mydestination = $myhostname $myhostname.$mydomain localhost.$mydomain localhost
sender_canonical_maps = static:$myhostname@$mydomain

Here is the tail end of the log file from overnight: (This happened over night while I was gone)

Apr  4 12:30:03 voip2 postfix/master[21006]: daemon started -- version 2.6.6, configuration /etc/postfix
Apr  4 12:59:25 voip2 postfix/postfix-script[22506]: stopping the Postfix mail system
Apr  4 12:59:25 voip2 postfix/master[21006]: terminating on signal 15
Apr  4 13:00:57 voip2 postfix/postfix-script[2374]: warning: not owned by root: /etc/postfix
Apr  4 13:00:57 voip2 postfix/postfix-script[2384]: warning: not owned by root: /etc/postfix/access
Apr  4 13:00:57 voip2 postfix/postfix-script[2385]: warning: not owned by root: /etc/postfix/canonical
Apr  4 13:00:57 voip2 postfix/postfix-script[2386]: warning: not owned by root: /etc/postfix/canonical.db
Apr  4 13:00:57 voip2 postfix/postfix-script[2387]: warning: not owned by root: /etc/postfix/generic
Apr  4 13:00:57 voip2 postfix/postfix-script[2388]: warning: not owned by root: /etc/postfix/header_checks
Apr  4 13:00:57 voip2 postfix/postfix-script[2389]: warning: not owned by root: /etc/postfix/main.cf
Apr  4 13:00:57 voip2 postfix/postfix-script[2390]: warning: not owned by root: /etc/postfix/master.cf
Apr  4 13:00:57 voip2 postfix/postfix-script[2391]: warning: not owned by root: /etc/postfix/relocated
Apr  4 13:00:57 voip2 postfix/postfix-script[2392]: warning: not owned by root: /etc/postfix/sasl
Apr  4 13:00:57 voip2 postfix/postfix-script[2393]: warning: not owned by root: /etc/postfix/sasl/sasl_passwd
Apr  4 13:00:57 voip2 postfix/postfix-script[2394]: warning: not owned by root: /etc/postfix/sasl/sasl_passwd.db
Apr  4 13:00:57 voip2 postfix/postfix-script[2395]: warning: not owned by root: /etc/postfix/transport
Apr  4 13:00:57 voip2 postfix/postfix-script[2396]: warning: not owned by root: /etc/postfix/virtual
Apr  4 13:00:57 voip2 postfix/postfix-script[2411]: starting the Postfix mail system
Apr  4 13:00:57 voip2 postfix/master[2412]: daemon started -- version 2.6.6, configuration /etc/postfix
Apr  4 13:21:15 voip2 postfix/postfix-script[8412]: refreshing the Postfix mail system
Apr  4 13:21:15 voip2 postfix/master[2412]: reload -- version 2.6.6, configuration /etc/postfix
Apr  5 00:00:01 voip2 postfix/sendmail[6683]: fatal: open /etc/postfix/main.cf: Permission denied
Apr  5 00:01:01 voip2 postfix/sendmail[6737]: fatal: open /etc/postfix/main.cf: Permission denied
Apr  5 00:06:24 voip2 postfix/sendmail[7139]: fatal: open /etc/postfix/main.cf: Permission denied
Apr  5 05:31:28 voip2 postfix/postfix-script[23065]: refreshing the Postfix mail system
Apr  5 05:31:28 voip2 postfix/master[2412]: reload -- version 2.6.6, configuration /etc/postfix
[root@voip2 log]#

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