@bajramia wrote:

Hi All,
I’m looking to add a survey for post call, so the queue can send the calls after they spoke to agents so the costumer would have some option like press 1, press 2, press 3, thank you.

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@bzaayer wrote:

This is a new installation of Freepbx 14.0.2.14 with Grandstream GXP-2170 phones (newest firmware in Endpoint). When we try to transfer anyone using the AttTrnf button on the phone it just blind transfers the call to the extension. I have tested transferring the call using *2 method and it works correctly. Our extension numbers are in the teens and twenty’s. I am not sure if this is causing the problem. I have other locations with the same phones and version of Freepbx and they do not have the issue. The only differences is the other locations have extensions in 100’s.
Any help in pointing me where to look for the problem would be appreciated.
Thanks,
Brian

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@markbhai wrote:

Hi All.

I don’t know why but I have a fatal error when trying to log into my server, the error is:
FATAL ERROR
DB Error: connect failed
Trace Back
/var/www/html/admin/libraries/db_connect.php:71 die_freepbx()
[0]: DB Error: connect failed

/var/www/html/admin/bootstrap.php:91 require_once()
[0]: /var/www/html/admin/libraries/db_connect.php

/etc/freepbx.conf:9 require_once()
[0]: /var/www/html/admin/bootstrap.php

/var/www/html/admin/config.php:99 include_once()
[0]: /etc/freepbx.conf

Under normal circumstances I would stop mysql, rename the mysql.sock file and restart mysql again, however…

I could not stop mysqld using the command service mysqld stop, so I ended up using the command pkill 20167 (which was the process ID of Mysql.

I managed to delete mysql.sock, but now I cannot restart using service mysqld start - I get a time out error.

I don’t know where I can find the log files (which I am sure will be the first bit of advice, so firstly I guess I would like someone to tell me where the mysql log files are.

Cheers

Mark,

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@steve_pbuk wrote:

When I click on “Latest Records” or filter via the months/days the results no longer appear. It just states “Loading, please wait…”

image.png1848x570 48.2 KB

The only thing I have done recently that may have affect this are:

1. I manually delete the recording archives via command line using the “rm” command.
2. I performed system/module updates the other day.

If I SSH to the server I can see the recording files in the monitor folder.

I’m on version: Call Recording Report 14.0.1.6

PBX Firmware:12.7.4-1804-1.sng7
PBX Service Pack:1.0.0.0
Current Asterisk Version: 13.19.1

Possible bug?

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@erike wrote:

Does FreePBX have a API or some other way for me to get the status of a queue/line? Right now we have for instance a tech support and a sales line and we sometimes close these for various reasons.

I want to be able to programmatically get the open/closed status to be able to send a notification in case we forget to open up the line again.

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@markbhai wrote:

Can anyone direct me to the best way to upgrade from:
FreePBX Framework - 12.0.74 to latest.; and
System Admin 2.10.0.77 to latest.

I try this and I get an error saying it cannot find the mirror or the scmooze server.

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@assos40 wrote:

Hi,
My freepbx is version 12.7.4-1804-1.sng7
My cdr call date is displayed like “Tue, Apr 10 2018 4:52 PM” on the cdr report.
In the cdr database the format is “2018-04-10 16:52:31”
Timezone is set in System Admin as Europe/Athens.
I need the call date like 10/4/2018 16:52"
Any help ,please
Thanks

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@pmit wrote:

Looking to secure voicemail access a bit more. Really hoping for SAML integration with UCP. We use Google 2FA and being able to point UCP to Google for SAML would really lock down our voicemail. That being said… Is there a way to prevent access to voicemail via phone and ONLY allow UCP access?

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@jasonmel wrote:

Hello,

User is going on vacation and forwards their extension to a different extension. Inbound call is forwarded correctly to the extension however if the person does not pickup the voicemail goes back to the original extension mailbox.

Once a call is forwarded, how can I ensure if the call is not picked up the voicemail goes to the forwarded extension?

Thanks,

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@tas1 wrote:

Hope someone can help me with this. I have a FreePBX install where most of the phones are on the same subnet as the server. Those all work as expected. There is one phone that is installed at a satellite office connecting over a VPN (handled by the routers). It is on a separate routed subnet. Once in a while there is a hiccup in the internet connection at the satellite office and the VPN get interrupted. When this happens the phone will not re-register once the VPN comes back up. I have tried rebooting the phone, and that does not help. The only thing that seems to work is is restarting dahdi & asterisk (through the web gui). I’m wondering if there’s a better/easier way. The PBX is 13.0.194.2 and the phone is a Grandstream GVX3240 with the latest firmware.

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@msnyder15 wrote:

Let me start with what is going on, and apologizing with how long this is, I just want to provide all I can upfront and be detailed in the issues and attempted fixes. We just moved buildings and decided to upgrade to the latest FreePBX since the version installed was from 2009. We set up a Dell Server with 12GB of RAM, running a minimal CentOS 7.4.1708 install. Software is FreePBX 14.2.0.10, Asterisk 15.2.0, and Dahdi 2.11.1+2.11.1. We are using a Digium 8 FXO AEX800 PCI-E Card with WCTDM24XXP module and VPMADT032 Hardware Echo Cancellation. We are running one DAHDI line into a hunt group of four total numbers and PJSIP extensions. When the phone goes down, the PJSIP lines still function with the ability to call extensions, but the incoming DAHDI line does not work until we run “fwconsole restart”. I have made it a point to call in 4-5 times a day, to check the status of the phones, which is not a solution for moving forward.

There are multiple issues that have presented itself since this install has gone live the past month. Almost occurs daily or every other day. Sometimes when people call in they are unable to hear the IVR. Other times people call in and it just rings and rings and rings without stopping and do not reach the IVR or default route set up by the IVR. Sometimes, people can call in, and the IVR answers, but does not accept any of the inputs on the IVR but if let alone, will reach the default route. The last of the reoccurring issues is when people call in, and our side answers, the calling in side cannot hear the answering side and it appears to be one-way sound.

We have been troubleshooting and scouring many of the WikiPages of help and many more help pages and feel like we have tried everything. We are hoping the community will be able to assist, before we attempt a rollback or reaching out to Digium.

The following have been error messages observed in dmesg which seem to be occurring prior to finding out the phones are down.

• Missed interrupt. Increasing latency to 5 ms in order to compensate.
• Unable to disable sw companding on echo cancellation channel 0 (reason 4)
• Unable to set SW Companding on channel 0 (reason 4)

Some of the things we have attempted to do to resolve the issue include, but are definitely not limited to:

• fwconsole restart fixes issues most of the time
• Changing from Software Echo Cancellation to Hardware
• Upgraded Server’s BIOS
• Latest Dahdi firmware
• Set relaxdtmf to ‘yes’
• Used fxotune to attempt to configure
• ‎Disabled the system frame buffer by adding nomodeset to the kernel boot line per the following knowledge base article: ‎https://support.digium.com/community/s/article/How-to-disable-the-Linux-frame-buffer-if-it-s-causing-problems
• Installed acpid.
• Made sure irqbalance was installed
• Installed OS updates and rebuilt the kernel module
• Ran the fxotune command again with a silence timeout of 2 seconds

https://www.voip-info.org/wiki/view/Asterisk+debugging
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
Issuing systemctl restart dahdi did not fix the issue. Only fwconsole restart did.

http://lists.digium.com/pipermail/asterisk-users/2009-August/236189.html

The only way we have been able to fully restore phone functionality would be to run “fwconsole restart”. This has also occasionally caused the FXO Port groups and Context to need to be reconfigured under DAHDI Config.

Any help or suggestions would be greatly appreciated!

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@Interlink wrote:

Hello, when trying to check for updates online I receive an error:

image.png1795x199 22.1 KB

I tried running yum update in cli and get the following:

Loaded plugins: fastestmirror, versionlock
Could not retrieve mirrorlist http://mirrorlist.sangoma.net/?release=7&arch=x86_64&repo=os&dist=$dist error was
12: Timeout on http://mirrorlist.sangoma.net/?release=7&arch=x86_64&repo=os&dist=$dist: (28, ‘Connection timed out after 30001 milliseconds’)

One of the configured repositories failed (Unknown),
and yum doesn’t have enough cached data to continue. At this point the only
safe thing yum can do is fail. There are a few ways to work “fix” this:

 1. Contact the upstream for the repository and get them to fix the problem.

 2. Reconfigure the baseurl/etc. for the repository, to point to a working
    upstream. This is most often useful if you are using a newer
    distribution release than is supported by the repository (and the
    packages for the previous distribution release still work).

 3. Run the command with the repository temporarily disabled
        yum --disablerepo=<repoid> ...

 4. Disable the repository permanently, so yum won't use it by default. Yum
    will then just ignore the repository until you permanently enable it
    again or use --enablerepo for temporary usage:

        yum-config-manager --disable <repoid>
    or
        subscription-manager repos --disable=<repoid>

 5. Configure the failing repository to be skipped, if it is unavailable.
    Note that yum will try to contact the repo. when it runs most commands,
    so will have to try and fail each time (and thus. yum will be be much
    slower). If it is a very temporary problem though, this is often a nice
    compromise:

        yum-config-manager --save --setopt=<repoid>.skip_if_unavailable=true

Cannot find a valid baseurl for repo: sng-base/7/x86_64

Current PBX Version:
14.0.2.17
Current System Version:
12.7.4-1803-1.sng7

Any help would be great!

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@subarutelescope wrote:

Successfully login to webportal, but click on any item under Module, returns a blank page.

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@toborgps wrote:

Hi Everyone!

I have fixed my phone issue and everything is working great now! I really would like to change the logos on my Cisco 7940 phones because well, I don’t love the logo… anyhow, I tried messing with my raspberry pi freepbx software for a tftp server, and was unable to connect to it, I can see the settings there, just not sure how to go beyond that or connecting to it. I appreciate all your help!

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@PitzKey wrote:

Hey guys,

I’m looking for a way to send an email to logged in agents only.

I was thinking of something, that me sending an email to the PBX (?) then the PBX should look for the agents logged in, poll their email address from their extension, and send it to them…?

In other words… The PBX should somehow receive data, and should pass it on to logged in agents email address that’s in the system already from their extension.

Any tips/help appreciated.

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@PitzKey wrote:

Hello all,

Day started by adding a new Google Voice account in motif module. I placed the email address as “gmailaddress” (without quotes) with the thought that the PBX will add the "@gmail.com".
However, once i added that account, asterisk crashed.

I tried restarting asterisk by running fwconsole stop and then fwconsole start, it returns:

Starting Asterisk...
[============================] 1 sec
Asterisk Started
Running Asterisk post from Endpoint module
Running Asterisk post from Pagingpro module
Running Asterisk post from Ucpnode module

But in fact the web gui says “cannot connect to asterisk”

I ran fwconsole ma upgradeall, then yum update it updated all modules and asterisk. It’s running now asterisk 13.19.1.
Rebooted.
Still no joy…

When i run service asterisk start i get the following messages:

/usr/sbin/safe_asterisk: line 171: 37408 Segmentation fault      (core dumped) nice -n $PRIORITY "${ASTSBINDIR}/asterisk" -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 37638 Segmentation fault      (core dumped) nice -n $PRIORITY "${ASTSBINDIR}/asterisk" -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
/usr/sbin/safe_asterisk: line 171: 37867 Segmentation fault      (core dumped) nice -n $PRIORITY "${ASTSBINDIR}/asterisk" -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
mv: cannot stat `/tmp/core.37869': No such file or directory
/usr/sbin/safe_asterisk: line 171: 38122 Segmentation fault      (core dumped) nice -n $PRIORITY "${ASTSBINDIR}/asterisk" -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}

So i ran service asterisk stop, and then fwconsole restart it says again that asterisk started.
Now i ran:

[root@localhost ~]# asterisk -vvvvvc
Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use 'asterisk -r' to connect.

[root@localhost ~]# asterisk -r
Asterisk 13.19.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 13.19.1 currently running on localhost (pid = 54370)
[2018-04-11 06:43:53] NOTICE[54370]: res_odbc.c:617 load_odbc_config: Registered ODBC class 'asteriskcdrdb' dsn->[MySQL-asteriskcdrdb]
[2018-04-11 06:43:53] NOTICE[54370]: res_odbc.c:1089 load_module: res_odbc loaded.
[2018-04-11 06:43:53] WARNING[54370]: pbx_config.c:1843 pbx_load_config: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 919 of /etc/asterisk/extensions_additional.conf
[2018-04-11 06:43:53] WARNING[54370]: pbx_config.c:1843 pbx_load_config: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 920 of /etc/asterisk/extensions_additional.conf
[2018-04-11 06:43:53] WARNING[54370]: presencestate.c:184 ast_presence_state_helper: No provider found for label CustomPresence
[2018-04-11 06:43:53] WARNING[54370]: presencestate.c:184 ast_presence_state_helper: No provider found for label CustomPresence
[2018-04-11 06:43:53] WARNING[54370]: presencestate.c:184 ast_presence_state_helper: No provider found for label CustomPresence
[2018-04-11 06:43:53] WARNING[54370]: presencestate.c:184 ast_presence_state_helper: No provider found for label CustomPresence
[2018-04-11 06:43:53] WARNING[54370]: presencestate.c:184 ast_presence_state_helper: No provider found for label CustomPresence
[2018-04-11 06:43:53] WARNING[54370]: presencestate.c:184 ast_presence_state_helper: No provider found for label CustomPresence
[2018-04-11 06:43:53] WARNING[54370]: presencestate.c:184 ast_presence_state_helper: No provider found for label CustomPresence
[2018-04-11 06:43:53] WARNING[54370]: logger:0 ***: Log queue threshold (1000) exceeded.  Discarding new messages.
[2018-04-11 06:43:53] WARNING[54376]: logger:0 ***: Logging resumed.  1501 messages discarded.
[2018-04-11 06:43:53] NOTICE[54370]: dnsmgr.c:494 do_reload: Managed DNS entries will be refreshed every 300 seconds.
[2018-04-11 06:43:53] ERROR[54370]: config_options.c:707 aco_process_config: Unable to load config file 'acl.conf'
[2018-04-11 06:43:53] NOTICE[54370]: cdr.c:4453 cdr_toggle_runtime_options: CDR simple logging enabled.
[2018-04-11 06:43:53] NOTICE[54370]: loader.c:1504 load_modules: 273 modules will be loaded.
[2018-04-11 06:43:53] ERROR[54370]: config_options.c:707 aco_process_config: Unable to load config file 'statsd.conf'
[2018-04-11 06:43:53] NOTICE[54370]: res_smdi.c:982 smdi_load: Unable to load config smdi.conf: SMDI disabled
[2018-04-11 06:43:53] NOTICE[54370]: res_smdi.c:1403 load_module: No SMDI interfaces are available to listen on, not starting SMDI listener.
[2018-04-11 06:43:53] WARNING[54370]: res_stun_monitor.c:340 load_config: Unable to load config res_stun_monitor.conf
[2018-04-11 06:43:53] ERROR[54370]: res_xmpp.c:3979 xmpp_client_config_post_apply: Jabber identity 'nhpublicphone2' could not be created for client 'gnhpublicphone2' - client not active
[2018-04-11 06:43:53] ERROR[54370]: config_options.c:707 aco_process_config: Unable to load config file 'hep.conf'
[2018-04-11 06:43:53] ERROR[54370]: res_calendar.c:1055 load_config: Unable to load config calendar.conf
[2018-04-11 06:43:53] ERROR[54370]: pbx_lua.c:1635 load_or_reload_lua_stuff: Error loading extensions.lua: cannot open '/etc/asterisk/extensions.lua' for reading: No such file or directory
[2018-04-11 06:43:53] WARNING[54370]: res_pktccops.c:1040 load_pktccops_config: Unable to load config file res_pktccops.conf
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_sips_contact.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_send_to_voicemail.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_dialog_info_body_generator.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_history.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_transport_management.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_path.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_xpidf_body_generator.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_phoneprov_provider.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_dlg_options.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'func_pjsip_contact.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_pidf_digium_body_supplement.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'func_pjsip_aor.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_mwi_body_generator.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_publish_asterisk.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_pidf_body_generator.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_hep_pjsip.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_empty_info.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_pidf_eyebeam_body_supplement.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54410]: res_xmpp.c:3787 xmpp_client_receive: Parsing failure: Hook returned an error.
[2018-04-11 06:43:53] WARNING[54410]: res_xmpp.c:3784 xmpp_client_receive: Parsing failure: Invalid XML.
[2018-04-11 06:43:53] WARNING[54410]: res_xmpp.c:3866 xmpp_client_thread: [gnhpublicphone1gmailcom] Socket read error
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_sips_contact.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_send_to_voicemail.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_dialog_info_body_generator.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_history.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_transport_management.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_path.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_xpidf_body_generator.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_phoneprov_provider.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_dlg_options.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'func_pjsip_contact.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_pidf_digium_body_supplement.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'func_pjsip_aor.so' could not be loaded.
[2018-04-11 06:43:53] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_mwi_body_generator.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_publish_asterisk.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_pidf_body_generator.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_hep_pjsip.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_empty_info.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_pidf_eyebeam_body_supplement.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_sips_contact.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_send_to_voicemail.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_dialog_info_body_generator.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_history.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_transport_management.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_path.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_xpidf_body_generator.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_phoneprov_provider.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_dlg_options.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'func_pjsip_contact.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_pidf_digium_body_supplement.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'func_pjsip_aor.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_mwi_body_generator.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_publish_asterisk.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_pidf_body_generator.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_hep_pjsip.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_empty_info.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_pidf_eyebeam_body_supplement.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_sips_contact.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_send_to_voicemail.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_dialog_info_body_generator.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_history.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_transport_management.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_path.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_xpidf_body_generator.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_phoneprov_provider.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_dlg_options.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'func_pjsip_contact.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_pidf_digium_body_supplement.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'func_pjsip_aor.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_mwi_body_generator.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_publish_asterisk.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_pidf_body_generator.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_hep_pjsip.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_empty_info.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_pidf_eyebeam_body_supplement.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'res_pjsip_sips_contact.so': /usr/lib64/asterisk/modules/res_pjsip_sips_contact.so: undefined symbol: ast_sip_register_service
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_sips_contact.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'res_pjsip_send_to_voicemail.so': /usr/lib64/asterisk/modules/res_pjsip_send_to_voicemail.so: undefined symbol: ast_sip_session_register_supplement
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_send_to_voicemail.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'res_pjsip_dialog_info_body_generator.so': /usr/lib64/asterisk/modules/res_pjsip_dialog_info_body_generator.so: undefined symbol: ast_sip_pubsub_register_body_generator
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_dialog_info_body_generator.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'res_pjsip_history.so': /usr/lib64/asterisk/modules/res_pjsip_history.so: undefined symbol: ast_sip_register_service
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_history.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'res_pjsip_transport_management.so': /usr/lib64/asterisk/modules/res_pjsip_transport_management.so: undefined symbol: ast_sip_register_service
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_transport_management.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'res_pjsip_path.so': /usr/lib64/asterisk/modules/res_pjsip_path.so: undefined symbol: ast_sip_unregister_supplement
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_path.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'res_pjsip_xpidf_body_generator.so': /usr/lib64/asterisk/modules/res_pjsip_xpidf_body_generator.so: undefined symbol: ast_sip_pubsub_register_body_generator
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_xpidf_body_generator.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'res_pjsip_phoneprov_provider.so': /usr/lib64/asterisk/modules/res_pjsip_phoneprov_provider.so: undefined symbol: ast_phoneprov_delete_extension
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_phoneprov_provider.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'res_pjsip_dlg_options.so': /usr/lib64/asterisk/modules/res_pjsip_dlg_options.so: undefined symbol: ast_sip_session_register_supplement
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_dlg_options.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'func_pjsip_contact.so': /usr/lib64/asterisk/modules/func_pjsip_contact.so: undefined symbol: ast_sip_get_contact_status_label
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'func_pjsip_contact.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'res_pjsip_pidf_digium_body_supplement.so': /usr/lib64/asterisk/modules/res_pjsip_pidf_digium_body_supplement.so: undefined symbol: ast_sip_pubsub_unregister_body_supplement
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_pidf_digium_body_supplement.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'func_pjsip_aor.so': /usr/lib64/asterisk/modules/func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'func_pjsip_aor.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'res_pjsip_mwi_body_generator.so': /usr/lib64/asterisk/modules/res_pjsip_mwi_body_generator.so: undefined symbol: ast_sip_pubsub_register_body_generator
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_mwi_body_generator.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'res_pjsip_publish_asterisk.so': /usr/lib64/asterisk/modules/res_pjsip_publish_asterisk.so: undefined symbol: ast_sip_publish_client_alloc_datastore
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_publish_asterisk.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'res_pjsip_pidf_body_generator.so': /usr/lib64/asterisk/modules/res_pjsip_pidf_body_generator.so: undefined symbol: ast_sip_pubsub_register_body_generator
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_pidf_body_generator.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'res_hep_pjsip.so': /usr/lib64/asterisk/modules/res_hep_pjsip.so: undefined symbol: ast_sip_register_service
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_hep_pjsip.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'res_pjsip_empty_info.so': /usr/lib64/asterisk/modules/res_pjsip_empty_info.so: undefined symbol: ast_sip_session_register_supplement
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_empty_info.so' could not be loaded.
[2018-04-11 06:43:54] WARNING[54370]: loader.c:586 load_dlopen: Error loading module 'res_pjsip_pidf_eyebeam_body_supplement.so': /usr/lib64/asterisk/modules/res_pjsip_pidf_eyebeam_body_supplement.so: undefined symbol: ast_sip_presence_exten_state_to_str
[2018-04-11 06:43:54] WARNING[54370]: loader.c:1190 load_resource: Module 'res_pjsip_pidf_eyebeam_body_supplement.so' could not be loaded.
[2018-04-11 06:43:54] ERROR[54370]: res_config_sqlite.c:733 load_config: Unable to load res_config_sqlite.conf
localhost*CLI>

EDIT: if reading the logs in pastebin is easier, here’s a link: https://pastebin.freepbx.org/view/917a0d47

Can anyone tell me what next?

Thanks

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@vespino wrote:

I have just finished installing a FreePBX server for one of my clients. All seems to be working fine and now they are discovering the possibilities I’m running into some questions I can’t answer. Hope I can find the answers here like so many times before! Here goes…

1. When a call comes in and is not answered they would like the call to be forwarded a cell phone. I can do this using Misc Destinations, but the CID shown on the cell phone is that of the office, where they would like this to be the original caller id. Is this possible?

2. If not possible: my client uses multiple numbers and based on the number they pick up the phone on behalf of a different company. I have currently set up 1 ring group and thus 1 Misc Destination. If I would like multiple numbers to be shown when forwarding, is the only way to do this to set up multiple ring groups?

3. When all else fails: is it possible to forward calls with the push of 1 button on their (Yealink desktop) phone? so 1 button to enable and disable the forwarding?

4. Then they have one extension of which they would like to force a certain CID. I have found the “Outbound CID” option in the extension module, but no matter what I enter, the CID is always set through the outbound routes.

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@steve_pbuk wrote:

Quick question.

Do I need to configure anything to enable an engaged tone if an extension is already on a call?

I have recently setup a new system using the FreePBX distro and Sangoma S500 phones and users are asking for the engaged tone to be added because currently the extension just rings.

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@mvogel4949 wrote:

I have two asterisk systems connected via IAX2 trunks. I am using the second system (FreePBX13 and Asterisk13-22, all modules current) to host the conference bridge. When the bridge gets over 23 people the audio starts to get really choppy. The Memory and Computer usage both are quite low so I don’t think the system is being overworked. Am I missing something in my IAX2 trunk setup?

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@Bradbpw wrote:

When my users login to UCP they get the following error:

“Unable to connect to the UCP Node Server because: ‘Error: xhr poll error’”

I have tried the following:
-I have checked my certificate. The same cert I have marked as default is also the cert installed under HTTPS settings
-I have restarted UCPnode

[root@freepbx asterisk]# fwconsole stop ucpnode
Running FreePBX shutdown...


[root@freepbx asterisk]# fwconsole start ucpnode
Running FreePBX startup...
Taking too long? Customize the chown command, See http://wiki.freepbx.org/display/FOP/FreePBX+Chown+Conf
Setting Permissions...
Setting specific permissions...
    3 [============================]
Finished setting permissions
Unable to run Pre-Asterisk hooks, because Asterisk is already running on PID 30095 and has been running for 1 hour, 6 minutes, 1 second
[root@freepbx asterisk]# 

• had installed Zulu server V3 beta as I was troubleshooting a Zulu issue. I uninstalled that from Module admin then ran “fwconsole ma downloadinstall zulu” from CLI to install the stable version (I assume that installed the stable version)

uninstalling XMPP and Conferences Pro as has been suggested DID fix the issue, but I need XMPP for Zulu. And I would like to keep Conferences Pro. Is there a way to fix my issue while keeping these two moduels (at least XMPP and Zulu)?

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