With Iax2 show peer [peer C] , both A and B site report "ACL: NO", while AtoB trunks have "ACL: YES"
This is the only difference with site C, anyway only A to C iax is unreachable... any other idea ?
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Suddenly, Iax peer is unreachable, trunk working one-way
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Zulu Connection dropped by remote peer
Here is the error
OUT > 2017-09-06 15:08 -04:00: [2017-09-06 15:08:17.444] [ERROR] console - Error: Table 'asterisk.smsrouting' doesn't exist_
Based on that error you needed to install sms to create that table.
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Can't upgrade 4 modules
I had the same issue. During the upgrade, the login shell for the "asterisk" user account was changed to /sbin/nologin, preventing that user from inputting commands.
In the CLI as root, enter the following: nano /etc/passwd
Arrow down to the asterisk line, and replace the "/sbin/nologin" with "/bin/bash" (without the quotes). Press Ctrl + X to exit, then Y to save. Try your installation again.
The user's line should read something like: asterisk:x:499:498::/var/lib/asterisk:/bin/bash
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Seperate failover destinations for internal and external calls
Hi There,
I have a client that requires their internal calls to failover to voicemail(or just cutoff) but when a client dials the DID to that user it fails over to the switchboard.
Is there an easy way to do this?
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Cannot Connect to Online Repository (mirror1.freepbx.org, mirror2.freepbx.org)
I also can now ping mirror 1
ping mirror1.freepbx.org
PING mirror1.freepbx.org (199.102.239.170) 56(84) bytes of data.
64 bytes from 199.102.239.170: icmp_seq=1 ttl=42 time=310 ms
64 bytes from 199.102.239.170: icmp_seq=2 ttl=42 time=310 ms
64 bytes from 199.102.239.170: icmp_seq=3 ttl=42 time=310 ms
64 bytes from 199.102.239.170: icmp_seq=4 ttl=42 time=310 ms
64 bytes from 199.102.239.170: icmp_seq=5 ttl=42 time=310 ms
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Cannot Connect to Online Repository (mirror1.freepbx.org, mirror2.freepbx.org)
I had manually upgrade the framework module
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PJSIP NAT AUDIO transport=transport-udp
Standard config for pjsip extensions works for this arrangement, you don't have to delve into conf files. What version of Asterisk, if not at 13.17.1, upgrade to current, there are recently fixed bugs with remote pjsip extensions.
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Sonidos es Castellano
Cordial Saludo.
La mejor forma de hacerlo es la siguiente:
Ingresar a Admin, Sound Languajes. Ahí debes descargar el sonido en español y habilitar en cada extensión este nuevo lenguaje.
Atentamente,
Yeison Manrique
Ing. Sistemas
Cel. +57 300 217 20 29
www.voipsystem.net.co
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Registration timed out trunk OVH
Hello,
I changed my server recently, and the registration of OVH trunks is denied.
siptrunk.ovh.net:5060 Y PHONENUMBER120 Request Sent
In the console, there are:
[2017-09-28 14:10:22] NOTICE [9992] chan_sip.c: - Registration for 'PHONENUMBER@siptrunk.ovh.net' timed out, trying again
(ping siptrunk.ovh.net is OK)
In verbose mode:
2017-09-28 13:56:02] VERBOSE [9992] chan_sip.c: REGISTER 11 headers, 0 lines
[2017-09-28 13:56:02] VERBOSE [9992] chan_sip.c: Reliably Transmitting (NAT) to SIPTRUNK.OVH.NET:5060:
REGISTER sip: siptrunk.ovh.net SIP / 2.0
Via: SIP / 2.0 / UDP MYIP; branch = xxx;
Max-Forwards: 70
From: sip: PHONENUMBER@siptrunk.ovh.net; tag = xxx
To: sip: PHONENUMBER@siptrunk.ovh.net
Call-ID: xxxxx
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.192.18 (13.17.2)
Expires: 120
Contact:
Content-Length: 0
[2017-09-28 13:56:02] VERBOSE [9992] chan_sip.c:
<--- SIP read from UDP: SIPTRUNK.OVH.NET: 5060 --->
SIP / 2.0 100 Trying
Call-ID: xxxxx
CSeq: 102 REGISTER
From: sip: PHONENUMBER@siptrunk.ovh.net; tag = xxxxx
To: sip: PHONENUMBER@siptrunk.ovh.net
Via: SIP / 2.0 / UDP MYIP: 5160; received = MYIP; rport = 5160; branch = xxx
Content-Length: 0
<------------->
[2017-09-28 13:56:02] VERBOSE [9992] chan_sip.c: --- (7 headers 0 lines) ---
[2017-09-28 13:56:02] VERBOSE [9992] chan_sip.c:
<--- SIP read from UDP: SIPTRUNK.OVH.NET: 5060 --->
SIP / 2.0 401 Unauthorized
Call-ID: xxxx
CSeq: 102 REGISTER
From: sip: PHONENUMBER@siptrunk.ovh.net; tag = xxxx
To: sip: PHONENUMBER@siptrunk.ovh.net; tag = xxxx
Via: SIP / 2.0 / UDP MYIP: 5160; received = MYIP; rport = 5160; branch = xxxx
WWW-Authenticate: Digest realm = "siptrunk.ovh.net", nonce = "xx", opaque = "xxxx", stale = false, algorithm = MD5
Server: Cirpack / v4.70 (gw_sip)
Content-Length: 0
peer details:
username=phonenumber
type=peer
secret=password
restrictcid=no
nat=yes
language=fr
insecure=port,invite
host=siptrunk.ovh.net
fromuser=phonenumber
dtmfmode=inband
defaultexpiry=1800
context=custom-get-did-ovh
canreinvite=no
amaflags=default
incoming Register String: phonenumber:password@siptrunk.ovh.net
extensions_custom.conf:
[custom-get-did-ovh]
exten => X.,1,Goto(from-trunk,${CUT(CUT(SIPHEADER(To),@,1),:,2)},1)
I have exactly the same config in the "trunk" menu of freepbx, sip.conf / sip_additional.conf, on the old and new server (copy paste, checked and re-checked).
The new server's IP is added in the OVH interface as allowed
siptrunk.ovh.net is in the firewall's whitelist (firewall off, same probleme)
The old server can register on this trunk, but not the new one ....
Do you have any ideas ?
Thank you
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Sonidos es Castellano
Hola yefreman, gracias por la respuesta pero no me refería a eso. La cuestion es que el idioma español que trae freepbx no es muy agradable para ciudadanos españoles, ya que tiene un fuerte acento angloamericano.
He creado el custom sound language, pero los sonidos de castellano de españa que he cargado son de la versión 1.4 y 1.8 de asterisk, y algunos sonidos se llaman diferente en la versión 11.
Alguna sugerencia para descargar sonidos en castellano para freepbx 13?
Saludos
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Sonidos es Castellano
Puedes usar la funcion que ya trae la version 13 de FreePBX para bajar los sonidos desde el menu Admin->Sound languages.
Tambien puedes bajar otras voces desde el siguiente link, yo las tengo funcionando correctamente en mi FreePBX 13
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Call Recordings Retention Period
I would like this feature, however, I'd also like a UCP > Call History > Controls Column > Delete icon next to the download icon. Is that an option currently achieved via permissions or something?
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Can't activate - fresh install
Could you solve the problem? I'm having the same issues
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How to delete call recordings?
UCP deletion and administrative retention control for the system would be great.
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[FIXED] FreePBX Administration GUI Blank Page
Hi Andrew, I just installed asterisk in a fresh CentOS 7 box using "wiki.freepbx.org/display/FOP/Installing+FreePBX+14+on+CentOS+7". Now IP/admin/config.php shows blank page.
amportal a dbug gives the following output. How to resolve this issue. I already run "rm -Rf /var/www/html/admin/modules/extensionroutes/" and restarted the httpd and asterisk service.
amportal a dbug
Please wait...
!!!!amportal is depreciated. Please use fwconsole!!!!
forwarding all commands to 'fwconsole'
+-----------------------+
| FreePBX Notifications |
+-----------------------+
Neither app_meetme nor app_confbridge is configured in Asterisk, conferencing, paging and other functionality will not work properly
The FreePBX project is collecting anonymous browser statistics using google analytics. These are used to focus development efforts based on real user input. All information is anonymous. You can disable this in Advanced Settings with the Browser Stats setting.
You are missing support for the following HTML5 codecs: mp3,m4a. To fully support HTML5 browser playback you will need to install programs that can not be distributed with FreePBX. If you'd like to install the binaries needed for these conversions click 'Resolve' in the lower left corner of this message. You can also safely ignore this message but browser playback might not work in your browser.
The default bind ports for FreePBX have changed. Please keep this is mind while configuring your devices. You can change this in SIP Settings. CHAN_PJSIP is: 5060, CHAN_SIP is: 5160
OUT > ==> /var/log/asterisk/freepbx_dbug <==
==> /var/log/httpd/error_log <==
[Thu Sep 28 18:30:59.848094 2017] [:error] [pid 30537] [client 192.168.0.125:58540] PHP Warning: include_once(): Failed opening '/etc/freepbx.conf' for inclusion (include_path='.:/usr/share/pear:/usr/share/php') in /var/www/html/admin/config.php on line 100
[Thu Sep 28 18:30:59.848125 2017] [:error] [pid 30537] [client 192.168.0.125:58540] PHP Fatal error: Class 'FreePBX' not found in /var/www/html/admin/config.php on line 110
[Thu Sep 28 18:30:59.848163 2017] [:error] [pid 30537] [client 192.168.0.125:58540] PHP Warning: Unknown: open(/var/lib/php/session/sess_3ad4j5dillqrgjbncncv693ln0, O_RDWR) failed: Permission denied (13) in Unknown on line 0
[Thu Sep 28 18:30:59.848175 2017] [:error] [pid 30537] [client 192.168.0.125:58540] PHP Warning: Unknown: Failed to write session data (files). Please verify that the current setting of session.save_path is correct (/var/lib/php/session) in Unknown on line 0
[Thu Sep 28 18:49:30.186954 2017] [authz_core:error] [pid 30538] [client 192.168.0.125:59046] AH01630: client denied by server configuration: /var/www/html/admin/index.html
[Thu Sep 28 18:49:30.190664 2017] [:error] [pid 30538] [client 192.168.0.125:59046] PHP Warning: include_once(/etc/freepbx.conf): failed to open stream: Permission denied in /var/www/html/admin/config.php on line 100
[Thu Sep 28 18:49:30.190690 2017] [:error] [pid 30538] [client 192.168.0.125:59046] PHP Warning: include_once(): Failed opening '/etc/freepbx.conf' for inclusion (include_path='.:/usr/share/pear:/usr/share/php') in /var/www/html/admin/config.php on line 100
[Thu Sep 28 18:49:30.190735 2017] [:error] [pid 30538] [client 192.168.0.125:59046] PHP Fatal error: Class 'FreePBX' not found in /var/www/html/admin/config.php on line 110
[Thu Sep 28 18:49:30.190772 2017] [:error] [pid 30538] [client 192.168.0.125:59046] PHP Warning: Unknown: open(/var/lib/php/session/sess_3ad4j5dillqrgjbncncv693ln0, O_RDWR) failed: Permission denied (13) in Unknown on line 0
[Thu Sep 28 18:49:30.190784 2017] [:error] [pid 30538] [client 192.168.0.125:59046] PHP Warning: Unknown: Failed to write session data (files). Please verify that the current setting of session.save_path is correct (/var/lib/php/session) in Unknown on line 0
==> /var/log/asterisk/freepbx.log <==
[2017-Sep-28 13:19:16] [INFO] (libraries/modulefunctions.class.php:1980) - Updating table outbound_route_patterns...Done
[2017-Sep-28 13:19:16] [INFO] (libraries/modulefunctions.class.php:1980) - Updating table outbound_route_sequence...Done
[2017-Sep-28 13:19:16] [INFO] (libraries/modulefunctions.class.php:1980) - Updating table outbound_route_trunks...Done
[2017-Sep-28 13:19:16] [INFO] (libraries/modulefunctions.class.php:1980) - Updating table outbound_routes...Done
[2017-Sep-28 13:19:16] [INFO] (libraries/modulefunctions.class.php:1980) - Updating table trunk_dialpatterns...Done
[2017-Sep-28 13:19:17] [INFO] (core/install.php:423) - Migrating pickup groups to named pickup groups
[2017-Sep-28 13:19:17] [INFO] (core/install.php:428) - Migrating call groups to named call groups
[2017-Sep-28 13:19:17] [INFO] (core/install.php:467) - Checking for possibly invalid emergency caller id fields..none found
[2017-Sep-28 13:19:17] [UPDATE] (libraries/modulefunctions.class.php:1996) - Module: core Updated to version 14.0.1.9
[2017-Sep-28 13:19:17] [INFO] (libraries/modulefunctions.class.php:2054) - Generating CSS...Done
ERR > tail: cannot open ‘/var/log/asterisk/freepbx_security.log’ for reading: No such file or directory
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FREEPBX 13: After module upgrade I have got permanent error at http://asterisk:9980/admin/config.php
(post withdrawn by author, will be automatically deleted in 24 hours unless flagged)
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Log in phone
Since I upgraded to FreePBX 14 (from RasPBX 13) and when I make a call the outgoing CID is shown in the display of my Yealink Phone. That's a great feature because it confirms I'm using the correct number when calling. So far, ok. But the log file also show CID: XXXXXXXXXX instead of the number I called which is not user friendly. I have to guess which number I would like to call again instead of just selecting the previous outgoing call and dial.
Is this something in FreePBX I could change?
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PJSIP NAT AUDIO transport=transport-udp
I am using freepbx 14 and asterisk 14.6.1.
Everything was working fine till I upgraded one of the modules (ring groups). I then got one way audio. Recovered from a backup image before the module update and working again. I now have the broken image on another machine and I debugged the sip packet sdp and noticed that the endpoint ip is offered instead of the asterisk asterisk ip. This is the current state and can be duplicated. Thanks for you help.
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Issue getting FollowMe to work to external
Looks like FollowMe isn't really working for any extension here.
- FreePBX 14
- FollowMe 14.0.1.13
- Core 14.0.1.3
Relevant lines from logs:
[2017-09-28 14:55:23] WARNING[32798][C-00000066] pbx_functions.c: Can't find trailing parenthesis for function 'IF(1?7749947661#:7749947661#'?
[2017-09-28 14:55:30] VERBOSE[32798][C-00000066] pbx.c: Executing [s@macro-dial:6] AGI("Local/FMGL-7749947661#@from-internal-000009bb;2", "dialparties.agi") in new stack
[2017-09-28 14:55:30] VERBOSE[32798][C-00000066] res_agi.c: dialparties.agi: Added extension xxx#) to extension map
[2017-09-28 14:55:30] VERBOSE[32798][C-00000066] res_agi.c: dialparties.agi: Extension xxx#) cf is disabled
[2017-09-28 14:55:30] VERBOSE[32798][C-00000066] res_agi.c: dialparties.agi: Extension xxx#) do not disturb is disabled
[2017-09-28 14:55:30] VERBOSE[32798][C-00000066] res_agi.c: dialparties.agi: Extension xxx#) has ExtensionState: 4
[2017-09-28 14:55:30] VERBOSE[32798][C-00000066] res_agi.c: dialparties.agi: Checking CW and CFB status for extension xxx#)
[2017-09-28 14:55:30] VERBOSE[32798][C-00000066] res_agi.c: dialparties.agi: Extension xxx#) is not available to be called
[2017-09-28 14:55:30] VERBOSE[32798][C-00000066] res_agi.c: <Local/FMGL-xxx#@from-internal-000009bb;2>AGI Script dialparties.agi completed, returning 0
[2017-09-28 14:55:30] VERBOSE[32798][C-00000066] pbx.c: Executing [s@macro-dial:7] NoOp("Local/FMGL-xxx#@from-internal-000009bb;2", "Returned from dialparties with no extensions to call and DIALSTATUS: NOANSWER") in new stackI see two issues:
* There's an error with that IF statement,
* The relevant part of dialparties.agi is checking for the extension state of an extension that has a trailing ')'
I re-ran with AGI debugging turned on and that is in fact the case:
<Local/FMGL-7749947661#@from-internal-000009c1;2>AGI Tx >> 200 result=1
<Local/FMGL-7749947661#@from-internal-000009c1;2>AGI Rx << GET VARIABLE EXTENSION_STATE(xxx#))
<Local/FMGL-7749947661#@from-internal-000009c1;2>AGI Tx >> 200 result=1 (UNKNOWN)
<Local/FMGL-7749947661#@from-internal-000009c1;2>AGI Rx << VERBOSE "EXTENSION_STATE: 4 (UNKNOWN)" 1
dialparties.agi: EXTENSION_STATE: 4 (UNKNOWN)
<Local/FMGL-7749947661#@from-internal-000009c1;2>AGI Tx >> 200 result=1
<Local/FMGL-7749947661#@from-internal-000009c1;2>AGI Rx << VERBOSE "Extension xxx#) has ExtensionState: 4" 1How should I resolve this issue? We are rarely at our desk phones and forwarding to cell phones is critical.
Thanks!
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Issue getting FollowMe to work to external
Hi!
You should update your modules... It was apparently fixed with core version 14.0.1.5...
re:
and the messages below it...
If 14.0.1.3 is the latest version you see it is possible you might have to switch to Edge for that module but I doubt this hasn't been published to the stable track by now...
Good luck and have a nice day!
Nick
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