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Asterisk 13.21.0
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Extension unable to call other extension
Something is still wrong with what you pasted, or the .conf file really is corrupted. The GotoIf statement should end with a ) but the last character is $
The $ is sometimes used to indicate that the line was cut off, so perhaps your editor or other processing between file and paste garbled it.
Are any module updates available that look relevant?
Unfortunately, I need to get to bed (it’s 02:50 here) but will try to answer tomorrow.
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Extension unable to call other extension
I’ll try and get another copy and no there aren’t any module updates. And no worries thanks for your help.
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FreePBX to FreePBX direct call without SIP trunk
We have a number of clients that we’ve deployed FreePBX for (say 30-40), and our firm also uses FreePBX for our own internal phone calls. Is there a way to make a web-based phone call from FreePBX-A directly over to FreePBX-B without traversing a SIP trunk or PSTN?
What I’m ultimately hoping to achieve is to program our client’s PBXs so that anyone who presses “00” on their phone system, will directly dial our HelpDesk (Ring Group 300 in our PBX). I should be able to place a call to 300@ourdomain.com, no?
I’m hoping to avoid programming direct IAX trunks - that sounds like the wrong tool for the job, right?
Any advice on this would be MUCH appreciated.
Thanks!
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Call recordings to include Announcements
There’s a setting under settings > advanced settings which allows you to record calls as soon as the call hits the Trunk. I forgot the name of the setting.
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Problem with CP-7962 not seeking the Config Server at FreePBX using Endpoint
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Call recordings to include Announcements
You are my hero.
I’ve pulled out [all] of my hair on this issue.
I can’t thank you enough!!!
SUCCESS!!
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Outbound call changes to hold music and drops
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FreePBX to FreePBX direct call without SIP trunk
The only way to do that without creating a direct sip or iax connection between the PBXs would be to setup sip uri calls. My guess it that it will be quicker and easier to go the way of the direct connection.
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Calls randomly hanging up
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Send Alert to phones if emergency call is placed
Thank you! Thats exactly what I was looking for.
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Queue Agents - how to include name or alias with extension
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Pjsip line feature broken in latest release?
Sorry for not providing more details initially.
I remembered that when looking at my VOIP provider Web interface I could see the registered endpoints and it showed some additional characters (assuming coming from the line option).
Now I see only:
|Device|Contact|
| FPBX-14.0.11(16.3.0) | 212.51.130.xxx:5060 |
|---|
I see the same registered device information for all 5 numbers (5 trunks defined)
This was definitely different compared to when I checked the last time and it was working.
Below some logs extracted from the FreePBX GUI.
What I’m getting when calling in is a message that the number is not in service.
Looking at the logs below this seems to come from Asterisk.
Here some logs from the FreePBX GUI:
[2019-06-19 04:02:01] Asterisk 16.3.0 built by mockbuild @ jenkins7 on a x86_64 running Linux on 2019-04-17 16:33:59 UTC
[2019-06-19 04:02:01] VERBOSE[43024] logger.c: Asterisk Queue Logger restarted
[2019-06-19 04:02:01] VERBOSE[43024] asterisk.c: Remote UNIX connection disconnected
[2019-06-19 14:36:11] VERBOSE[118433] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘212.51.130.xxx’
[2019-06-19 14:36:11] VERBOSE[13662][C-00000002] pbx.c: Executing [s@from-pstn:1] NoOp(“PJSIP/Home610-00000001”, “No DID or CID Match”) in new stack
[2019-06-19 14:36:11] VERBOSE[13662][C-00000002] pbx.c: Executing [s@from-pstn:2] Answer(“PJSIP/Home610-00000001”, “”) in new stack
[2019-06-19 14:36:12] WARNING[13662][C-00000002] chan_sip.c: This function can only be used on SIP channels.
[2019-06-19 14:36:12] VERBOSE[13662][C-00000002] pbx.c: Executing [s@from-pstn:3] Log(“PJSIP/Home610-00000001”, "WARNING,Friendly Scanner from ") in new stack
[2019-06-19 14:36:12] WARNING[13662][C-00000002] Ext. s: Friendly Scanner from
[2019-06-19 14:36:12] VERBOSE[13662][C-00000002] pbx.c: Executing [s@from-pstn:4] Wait(“PJSIP/Home610-00000001”, “2”) in new stack
[2019-06-19 14:36:14] VERBOSE[13662][C-00000002] pbx.c: Executing [s@from-pstn:5] Playback(“PJSIP/Home610-00000001”, “ss-noservice”) in new stack
[2019-06-19 14:36:14] VERBOSE[13662][C-00000002] file.c: <PJSIP/Home610-00000001> Playing ‘ss-noservice.alaw’ (language ‘en’)
[2019-06-19 14:36:15] VERBOSE[13662][C-00000002] pbx.c: Executing [h@from-pstn:1] Macro(“PJSIP/Home610-00000001”, “hangupcall,”) in new stack
[2019-06-19 14:36:15] VERBOSE[13662][C-00000002] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/Home610-00000001”, “1?theend”) in new stack
[2019-06-19 14:36:15] VERBOSE[13662][C-00000002] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2019-06-19 14:36:15] VERBOSE[13662][C-00000002] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/Home610-00000001”, “0?Set(CDR(recordingfile)=)”) in new stack
[2019-06-19 14:36:15] VERBOSE[13662][C-00000002] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/Home610-00000001”, " montior file= ") in new stack
[2019-06-19 14:36:15] VERBOSE[13662][C-00000002] pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“PJSIP/Home610-00000001”, “1?skipagi”) in new stack
[2019-06-19 14:36:15] VERBOSE[13662][C-00000002] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2019-06-19 14:36:15] VERBOSE[13662][C-00000002] pbx.c: Executing [s@macro-hangupcall:7] Hangup(“PJSIP/Home610-00000001”, “”) in new stack
[2019-06-19 14:36:15] VERBOSE[13662][C-00000002] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/Home610-00000001’ in macro ‘hangupcall’
[2019-06-19 14:36:15] VERBOSE[13662][C-00000002] pbx.c: Spawn extension (from-pstn, h, 1) exited non-zero on ‘PJSIP/Home610-00000001’
[2019-06-19 22:55:28] VERBOSE[15297] asterisk.c: Remote UNIX connection
[2019-06-19 22:55:40] VERBOSE[15297] asterisk.c: Remote UNIX connection
[2019-06-19 22:55:40] VERBOSE[93703] asterisk.c: Remote UNIX connection disconnected
I’m also looking through other logs and I see some errors in ucp_err.log
2019-06-19 00:24 +02:00: { Error: MySQL server has gone away code: 2006 }
There was an error with MySQL Connection
2019-06-19 08:24 +02:00: { Error: MySQL server has gone away code: 2006 }
There was an error with MySQL Connection
2019-06-19 16:24 +02:00: { Error: MySQL server has gone away code: 2006 }
There was an error with MySQL Connection
Asterisk version:
Asterisk 16.3.0, Copyright © 1999 - 2018, Digium, Inc. and others.
Also some error recently after a reboot in messages:
Jun 18 00:24:10 freepbx php: Unable to see safemode in services… Sleeping 5 seconds and retrying
Jun 18 00:24:10 freepbx systemd: Stopping Fail2Ban Service…
Jun 18 00:24:15 freepbx php: Unparseable output from getservices - [“Exception: Asterisk is not connected in file /var/www/html/admin/libraries/php-asmanager.php on line 242”,“Stack trace:”," 1. Exception->() /var/www/html/admin/libraries/php-asmanager.php:242"," 2. AGI_AsteriskManager->send_request() /var/www/html/admin/modules/firewall/Smart.class.php:447"," 3. FreePBX\modules\Firewall\Smart->getPjsipContacts() /var/www/html/admin/modules/firewall/Smart.class.php:437"," 4. FreePBX\modules\Firewall\Smart->getRegistrations() /var/www/html/admin/modules/firewall/Smart.class.php:69"," 5. FreePBX\modules\Firewall\Smart->getAllPorts() /var/www/html/admin/modules/firewall/Firewall.class.php:1026"," 6. FreePBX\modules\Firewall->getSmartPorts() /var/www/html/admin/modules/firewall/bin/getservices:22"] - returned 1
Jun 18 00:24:15 freepbx php: Unable to see safemode in services… Sleeping 5 seconds and retrying
Jun 18 00:24:16 freepbx fail2ban-client: Shutdown successful
Jun 18 00:24:16 freepbx systemd: Stopped Fail2Ban Service.
Jun 18 00:24:16 freepbx systemd: Starting Fail2Ban Service…
Jun 18 00:24:16 freepbx fail2ban-client: 2019-06-18 00:24:16,955 fail2ban.server [15175]: INFO Starting Fail2ban v0.8.14
Jun 18 00:24:16 freepbx fail2ban-client: 2019-06-18 00:24:16,955 fail2ban.server [15175]: INFO Starting in daemon mode
Jun 18 00:24:18 freepbx systemd: Started Fail2Ban Service.
Jun 18 00:24:20 freepbx systemd: Reloading.
Jun 18 00:24:21 freepbx php: Unparseable output from getservices - [“Exception: Asterisk is not connected in file /var/www/html/admin/libraries/php-asmanager.php on line 242”,“Stack trace:”," 1. Exception->() /var/www/html/admin/libraries/php-asmanager.php:242"," 2. AGI_AsteriskManager->send_request() /var/www/html/admin/modules/firewall/Smart.class.php:447"," 3. FreePBX\modules\Firewall\Smart->getPjsipContacts() /var/www/html/admin/modules/firewall/Smart.class.php:437"," 4. FreePBX\modules\Firewall\Smart->getRegistrations() /var/www/html/admin/modules/firewall/Smart.class.php:69"," 5. FreePBX\modules\Firewall\Smart->getAllPorts() /var/www/html/admin/modules/firewall/Firewall.class.php:1026"," 6. FreePBX\modules\Firewall->getSmartPorts() /var/www/html/admin/modules/firewall/bin/getservices:22"] - returned 1
Jun 18 00:24:21 freepbx php: Unable to see safemode in services… Sleeping 5 seconds and retrying
Jun 18 00:24:26 freepbx wall[15434]: wall: user root broadcasted 2 lines (34 chars)
Jun 18 00:24:26 freepbx php: Wall: ‘Firewall service now starting.#012#012’ returned 0
Jun 18 00:24:26 freepbx kernel: nf_conntrack version 0.5.0 (16384 buckets, 65536 max)
Basically I have to manually start asterisk with
fwconsole start
So something seems to be wrong with my system
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Pjsip line feature broken in latest release?
There seem to be more errors in the whole system.
Just did a reload after a module upgrade and I see errors like:
[2019-06-19 23:44:19] WARNING[118777]: pbx.c:7089 add_priority: Extension ‘s’ priority 13 in ‘macro-user-logon’, label ‘gotpass’ already in use at priority 9
[2019-06-19 23:44:19] ERROR[118777]: pbx.c:7313 ast_add_extension2_lockopt: You have to be kidding-- add exten ‘’ to context clean? Figure out a name and call me back. Action ignored.
[2019-06-19 23:44:19] WARNING[118777]: pbx_config.c:1891 pbx_load_config: Unable to register extension at line 4795 of /etc/asterisk/extensions_additional.conf
[2019-06-19 23:44:19] ERROR[118777]: pbx.c:7313 ast_add_extension2_lockopt: You have to be kidding-- add exten ‘’ to context clean? Figure out a name and call me back. Action ignored.
[2019-06-19 23:44:19] WARNING[118777]: pbx_config.c:1891 pbx_load_config: Unable to register extension at line 4796 of /etc/asterisk/extensions_additional.conf
[2019-06-19 23:44:19] ERROR[118777]: pbx.c:7313 ast_add_extension2_lockopt: You have to be kidding-- add exten ‘’ to context clean? Figure out a name and call me back. Action ignored.
[2019-06-19 23:44:19] WARNING[118777]: pbx_config.c:1891 pbx_load_config: Unable to register extension at line 4797 of /etc/asterisk/extensions_additional.conf
[2019-06-19 23:44:19] ERROR[118777]: pbx.c:7313 ast_add_extension2_lockopt: You have to be kidding-- add exten ‘’ to context mini-bar? Figure out a name and call me back. Action ignored.
[2019-06-19 23:44:19] ERROR[37115]: res_pjsip_config_wizard.c:1091 object_type_loaded_observer: Unable to load config file ‘pjsip_wizard.conf’
[2019-06-19 23:44:19] ERROR[37115]: res_pjsip_config_wizard.c:1091 object_type_loaded_observer: Unable to load config file ‘pjsip_wizard.conf’
[2019-06-19 23:44:19] WARNING[37115]: res_pjsip/config_transport.c:653 transport_apply: TOS and COS values ignored for websocket transport
[2019-06-19 23:44:19] WARNING[37115]: res_pjsip/config_transport.c:653 transport_apply: TOS and COS values ignored for websocket transport
[2019-06-19 23:44:19] WARNING[37115]: res_pjsip/config_transport.c:653 transport_apply: TOS and COS values ignored for websocket transport
[2019-06-19 23:44:19] WARNING[37115]: res_pjsip/config_transport.c:653 transport_apply: TOS and COS values ignored for websocket transport
[2019-06-19 23:44:19] ERROR[37115]: res_pjsip_config_wizard.c:1091 object_type_loaded_observer: Unable to load config file ‘pjsip_wizard.conf’
[2019-06-19 23:44:19] ERROR[37115]: res_pjsip_config_wizard.c:1091 object_type_loaded_observer: Unable to load config file ‘pjsip_wizard.conf’
[2019-06-19 23:44:19] NOTICE[37115]: sorcery.c:1334 sorcery_object_load: Type ‘system’ is not reloadable, maintaining previous values
[2019-06-19 23:44:19] ERROR[37115]: res_pjsip_config_wizard.c:1091 object_type_loaded_observer: Unable to load config file ‘pjsip_wizard.conf’
– Reloading module ‘res_pjsip_authenticator_digest.so’ (PJSIP authentication resource)
– Reloading module ‘res_resolver_unbound.so’ (Unbound DNS Resolver Support)
[2019-06-19 23:44:19] ERROR[118777]: config_options.c:710 aco_process_config: Unable to load config file ‘resolver_unbound.conf’
– Reloading module ‘res_pjsip_endpoint_identifier_ip.so’ (PJSIP IP endpoint identifier)
[2019-06-19 23:44:19] ERROR[118777]: res_pjsip_config_wizard.c:1091 object_type_loaded_observer: Unable to load config file ‘pjsip_wizard.conf’
[2019-06-19 23:44:19] ERROR[118777]: res_pjsip_config_wizard.c:1091 object_type_loaded_observer: Unable to load config file ‘pjsip_wizard.conf’
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How to configure phone directory on Asterisk 13
Hello. I’ve just installed Asterisk 13, but can’t get the directory active so my Polycom phones could get it on them, what I need to do?, I know how to do it on Asterisk 11, but version 6.211.65.
Please help.
Thanks in advance.
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Endpoint Manager - Yealink Common Configuration files
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Import FreePBX distro on GCE problem
This iso image should work
https://downloads.freepbxdistro.org/ISO/SNG7-FPBX-64bit-1805-2.iso
I tested this iso on a local VM that is VHD. (VMDK
may work as well https://cloud.google.com/compute/docs/images/importing-virtual-disks) wit fixed size.
Make sure you create a service account and download the JSON file https://cloud.google.com/compute/docs/vm-migration/using-cloud-endure
Here is some more guide https://www.youtube.com/watch?time_continue=39&v=R9JDCRCro-s
If you manage to have the VM instance up and running I recommend making Snapshot. I had issue updating the FreePBX system from the web interface. I recommend loging into your system via SSH and run the update from there.
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All circuits are busy and 401 unauthorized
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Sysconfig issues
I don’t know whether to laugh with you or not. When I first saw this, I thought Freepbx is joking and better humorous than my Bixby which I sometimes make it to joke or converse with me.
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Second time Freepbx has just stopped
A couple weeks ago my FreePBX machine was unresponsive, I couldn’t connect in any way, GUI didn’t load and SSH wouldn’t answer yet the HD led would flash occasionally like it was alive. Hard reboot once followed by a graceful reboot brought it back to life.
Now a couple weeks later and the GUI opens fine but says Asterisk isn’t responding in the right corner.
SSH in and try Asterisk -r and get no response. Next tried reboot now and it rebooted and everything came back up.
I though maybe the first failure was because its in a hot room but today is not so hot, Any thoughts on what could make Asterisk just quit and are there any maintenance commands I should run to check on filesystem or disk health? The Dashboard says its fine and updated modules and system fully but I’m a Linux idiot so I’m a little concerned to google an answer to this for fear of getting bad advice.
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