@posi211 wrote:
I'm running the FreePBX distro FreePBX 13.0.192.8
Phones are Sangoma connected using PJSIP.
I use EPM to program the phones.
In the parking lot default lot I selected Pickup Courtesy Tone to NONE
Now I test it and the customer is saying they still get the tones on the extension side when parking the call.Does anyone want to comment on this one?
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Participants: 1
@ABBA wrote:
I tried twice to install freepbx 13 on debian jessie, after add some extensions, reload failed always.
after "fwconsole reload"
/root$ fwconsole reload
Reloading FreePBX
Error(s) have occured, the following is the retrieve_conf output:
exit: 1
Exception: Unable to connect to Asterisk Manager from /var/lib/asterisk/bin/retrieve_conf, aborting in file /var/lib/asterisk/bin/retrieve_conf on line 11
Stack trace:
1. Exception->() /var/lib/asterisk/bin/retrieve_conf:11Another issue: PJSIP extensions disappeared from extension menu. I am sure I installed PJSIP and it disappeared after reboot today.
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@astrakid wrote:
hi,
i am using freepbx 14 on a vm. everything is running fine.
after todays update via "yum update" and a reboot i get a kernel panic. when rebooting with a former kernel it works fine.running kernel:
Linux freepbx.sangoma.local 3.10.0-514.16.1.el7.x86_64 #1 SMP Wed Apr 12 15:04:24 UTC 2017 x86_64 x86_64 x86_64 GNU/Linuxnot working:
initramfs-3.10.0-514.10.2.el7.x86_64.imgdoes anyone experienced something similar?
i can reproduce the issue. botting the newest kernel -> kernel panic
booting the former one -> freepbx starts fine.regards,
astrakid
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@adventurer wrote:
I've spent a lot of time trying to troubleshoot this, but still can't get it to work, so I figured it's time to get help from experts in this awesome community.
The FreePBX (10.13.66.20) server is hosted on Vultr. It works perfectly, except for a group of users who are in one office behind a Cisco firewall. Because these users can't install a softphone (corporate IT won't allow installation of new software), we have these users connecting via WebRTC on Firefox. From outside their network, the same PC using WebRTC works perfectly. However, when the users are behind their Cisco firewall, we haven't been able to get WebRTC to work. We get the green phone icon and can make a call, for example *60, but we don't hear any audio.
Here are the outbound ports that we've had their IT open up on their Cisco firewall:
Outbound ports from their network to the FreePBX server:
service-object tcp destination eq www
service-object tcp destination eq 81
service-object tcp destination eq https
service-object tcp destination eq 8083
service-object tcp destination eq 8088
service-object tcp destination eq 8089
service-object udp destination eq 81
service-object udp destination eq 8083
service-object udp destination eq 8088
service-object udp destination eq 8089
service-object udp destination range 10000 20000Outbound ports from their network to stun.counterpath.com:
service-object udp destination eq 3478I know that those are more ports than we really need, but we've opened up a lot more ports in order to troubleshoot. I think we really just need outbound 443 TCP, 8089 TCP, and 10000-20000 UDP to the FreePBX server, and 3478 UDP to the STUN server - is that right?
What are we missing here in order to get WebRTC to work behind their Cisco firewall? Thank you for your help!
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@BARDINET wrote:
Hi,
What happen if you leave office without having activated call forward from your device to your mobile , and dont have acces to the panel of FreePBX and no internet connection?That's why i'm looking for an elegant way to activate, from my mobile, a call formard for my device or a login of my device into a queue.
I can use Call Flow Control, and activate *280 ( or other) as a destination, that's a way, but not elegant.
The ideal solution for me could be a destination like *21 ( call forward) + Device to forward + forward destination .
Does anyone have an idea ?
Best regards,
Lionel.
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@kierank wrote:
Hi I ran the updates to version 10.13.66-20. All seemed well. When I click the red apply config button I now get the error listed above. When I click more info I get this.
exit: 255 Unable to continue. SQLSTATE[42S22]: Column not found: 1054 Unknown column 'writetimeout' in 'field list' in /var/www/html/admin/libraries/BMO/Database.class.php on line 239 #0 /var/www/html/admin/libraries/BMO/Database.class.php(239): PDO->query('INSERT INTO man...') #1 /var/www/html/admin/libraries/BMO/Database.class.php(208): FreePBX\Database->sql_query('INSERT INTO man...') #2 /var/www/html/admin/libraries/DB.class.php(235): FreePBX\Database->sql('INSERT INTO man...', 'query', 3) #3 /var/www/html/admin/libraries/sql.functions.php(29): DB->sql('INSERT INTO man...', 'query') #4 /var/www/html/admin/modules/manager/functions.inc.php(212): sql('INSERT INTO man...') #5 /var/www/html/admin/modules/cxpanel/functions.inc.php(1632): manager_add('cxpanel', 'cxmanager*con', '0.0.0.0/0.0.0.0', '127.0.0.1/255.2...', 'system,call,log...', 'system,call,log...') #6 /var/www/html/admin/modules/cxpanel/functions.inc.php(320): cxpanel_create_manager() #7 /var/www/html/admin/libraries/BMO/DialplanHooks.class.php(95): cxpanel_get_config('asterisk') #8 /var/lib/asterisk/bin/retrieve_conf(864): FreePBX\DialplanHooks->processHooks('asterisk', Array) #9 {main}Any ideas on how to fix it. Thanks.
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@JessicaRabbit wrote:
Following the recent update to OSS EPM, I would like to know the status of its compatibility with v13 and the upcoming v14 based on experience or testing. Thank You.
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@BCS_Inc_WORMA wrote:
When viewing history on phone, if I add a number to my Contacts, whre is that number stored. How do i access my contacts for that phone?
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@BCS_Inc_WORMA wrote:
The Gpickup key does not work when I assign to any one of the horizontal softkeys under the display. Dialing the *8# does work.Any ideas?
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@minkhant wrote:
Hi everyone,
After modules update I tried to click Apply Config and I had error. Please, look at attached picture.How can I solve this problem?
Help me Plz.
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Participants: 1
@dcitelecom wrote:
We have a customer who insists to use his old Nortel PBX system so we provide a SIP line via Linksys SPA 2102 adapters to his office and he connects it to his PBX like a POTS line.
The setup works ok, and we are not going with the customer about it, but now there is a problem when he wants to retrieve his PBX voice mail from outside the office. To access the Nortel voice mail he dials * * which is supposed to take him to voice mail. Instead it takes him to the company directory. Our log shows we are transmitting the ** digits but his PBX does not react to it. When they connect a real POTS line to the PBX, ** works.
Anyone have an idea of why this is happening?
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Participants: 2
@munozj wrote:
Using the instructions on: https://wiki.freepbx.org/display/FOP/Installing+FreePBX+13+on+CentOS+6
I installed this on CentOS 6.9 in Azure.
I tried several times and each time after the last step i'm able to reach the GUI and Asterisk without any issue but if I issue a reload, asterisk doesn't start and I have a big red box, "Cannot connect to asterisk"/etc/var/messages shows this error repeating:
Jun 27 16:50:37 azsc-voip-2 kernel: asterisk[4376]: segfault at 48 ip 00007fbbdda27531 sp 00007ffc5ddeef40 error 4 in res_ari.so[7fbbdda23000+d000] Jun 27 16:50:37 azsc-voip-2 abrt[4446]: Not saving repeating crash in '/usr/sbin/asterisk' Jun 27 16:50:43 azsc-voip-2 abrt[4446]: Saved core dump of pid 4376 to core.4376 at /tmp (102215680 bytes) Jun 27 16:50:49 azsc-voip-2 kernel: asterisk[4454]: segfault at 48 ip 00007faa46307531 sp 00007ffea64e2ff0 error 4 in res_ari.so[7faa46303000+d000] Jun 27 16:51:00 azsc-voip-2 abrt[4524]: Saved core dump of pid 4454 (/usr/sbin/asterisk) to /var/spool/abrt/ccpp-2017-06-27-16:50:49-4454 (102211584 bytes) Jun 27 16:51:00 azsc-voip-2 abrtd: Directory 'ccpp-2017-06-27-16:50:49-4454' creation detected Jun 27 16:51:00 azsc-voip-2 abrtd: Executable '/usr/sbin/asterisk' doesn't belong to any package and ProcessUnpackaged is set to 'no' Jun 27 16:51:00 azsc-voip-2 abrtd: 'post-create' on '/var/spool/abrt/ccpp-2017-06-27-16:50:49-4454' exited with 1 Jun 27 16:51:00 azsc-voip-2 abrtd: Deleting problem directory '/var/spool/abrt/ccpp-2017-06-27-16:50:49-4454'I tried stopping and restarting the service manually but it never returned "OK" after starting the service back up. Not sure what I might be doing wrong.
I have an identical VM built that appears to be running FreePBX 13 well.
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@jessy5765 wrote:
The phone system I am installing today I have 2 users that want to have the same extension at 2 different locations. So the internal LAN will be one phone and a Remote Phone will be another. I believe I can manually program this but I want to have Endpoint Manager handle this. is it possible?
I am using PJSIP so i know multiregistration is possible.
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Participants: 3
@RedHatter wrote:
I used the wiki guide "Installing FreePBX 14 on CentOS 7" but when I go to the FreePBX dashboard I see 'Cannot Connect To Asterisk' in the upper right hand corner. When I try and start Asterisk manually I get a bunch of errors then a seg fault. Do I need to configure Asterisk or something?
[root@hermes-voip ~]# /usr/sbin/asterisk -f -C /etc/asterisk/asterisk.conf PBX UUID: e2d6cb1c-3acd-4619-a2b6-776055c689ee Unable to load config file 'stasis.conf' Could not load Stasis configuration; using defaults [Jun 27 19:37:49] ERROR[3541]: logger.c:1823 init_logger: Errors detected in logger.conf. Default console logging is bei ng used. [Jun 27 19:37:49] NOTICE[3541]: loader.c:1367 load_modules: 2 modules will be loaded. [Jun 27 19:37:49] WARNING[3541]: loader.c:526 load_dynamic_module: Error loading module 'chan_local.so': /usr/lib64/aster isk/modules/chan_local.so: cannot open shared object file: No such file or directory [Jun 27 19:37:49] ERROR[3541]: config_options.c:655 aco_process_config: Unable to load config file 'features.conf' [Jun 27 19:37:49] NOTICE[3541]: features_config.c:1857 load_config: Could not load features config; using defaults [Jun 27 19:37:49] NOTICE[3541]: dnsmgr.c:494 do_reload: Managed DNS entries will be refreshed every 300 seconds. [Jun 27 19:37:49] ERROR[3541]: config_options.c:655 aco_process_config: Unable to load config file 'acl.conf' [Jun 27 19:37:49] ERROR[3541]: config_options.c:655 aco_process_config: Unable to load config file 'cdr.conf' [Jun 27 19:37:49] NOTICE[3541]: cdr.c:4110 process_config: Failed to process CDR configuration; using defaults [Jun 27 19:37:49] NOTICE[3541]: cdr.c:4213 cdr_toggle_runtime_options: CDR simple logging enabled. [Jun 27 19:37:49] ERROR[3541]: config_options.c:655 aco_process_config: Unable to load config file 'udptl.conf' [Jun 27 19:37:49] NOTICE[3541]: udptl.c:1324 __ast_udptl_reload: Could not load udptl config; using defaults [Jun 27 19:37:49] ERROR[3541]: config_options.c:655 aco_process_config: Unable to load config file 'cel.conf' [Jun 27 19:37:49] NOTICE[3541]: cel.c:1776 ast_cel_engine_init: Failed to process CEL configuration; using defaults bSIP channel loading... zSegmentation fault (core dumped)
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@fetoa wrote:
Hi,
I have System Admin Pro allready working an activates. I use dynamic DNS service and in it works, but 15 minutes is too long for me. Is it possible to force interval to one minute?
Regards
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Participants: 2
@rickbeare wrote:
I have FreePBX13.0.192.8, Asterisk11.23.1, with a Quintum Tenor AX 24FXO, Phones
5- Sangoma 500, and Digium D40. Cisco SG300-28PP 28-Port Gigabit PoE. PBX running on VMware.I built the whole system in our lab before delivery to the site, and either we did not notice the echo or it was not there. The only difference is the POTS lines which are actually both from ATT in our lab and on remote site. I have replicated the problem back to my lab via VPN to a Digium D40 phone from the remote site, and it has a VERY LOAD ECHO on the voip user side. When calling internal there is no ECHO. I have the Audio Codec set to G711Mu and A-law
Looking for suggestions on where to go look, I built an exact setup again here in my lab, with every thing the same except the Cisco switch and I only have digium phones in the lab, the remote site has Sangoma Phones, same Tenor for POTS, and I do not have an ECHO?
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@ChrisMaverley wrote:
Hi Folks,
I am baffled. The server was behind a firewall. I was alerted by our peer that they were seeing several calls to international destinations and they blocked the trunk. By the time I read the email and took action all looked ok. Now this baffles me there are no:ssh access records.
Call records.
Call recordings.
Evidence of files being tampered.Now I should mention I was working on a project to move our VoIP to AWS platform, but I had AWS firewall rules in place. Has anyone seen this type of clever hack before and have any info on how they can make calls with no record of such on the server?
Thanks,
Chris.
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Participants: 3
@ronandann wrote:
We purchased a Sangoma S700 with expansion module for our switchboard operator. I am trying to get it setup in the Commercial Endpoint Manager with the setup she was used to before. She was using an Aastra 6739i. Before we had some line keys setup to transfer to a direct extension but when I choose transfer for the S700 in Endpoint Manager, the value goes away so I am unable to put in the extension to transfer too. Is there another way of doing this?
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@VoIPTek wrote:
Hello All,
Running the latest release and when I went to the admin interface I received the error:
PHP Parse error: syntax error, unexpected $end in /var/www/html/admin/modules/directory/functions.inc.php on line 537Also, the firewall is complaining with:
Firewall was unable to connect to MySQL after 30 seconds.
Check Database!This is a fresh install about 2 weeks old.
Yum shows everything i current, so no updates to potentially fix this.
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Participants: 2
@trchavez99 wrote:
Anyone know of a good desktop fax app that will connect via SIP to FreePBX?
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