Ok. Can you please stop opening multiple tickets and multiple threads with the same information. You are only going to hurt yourself as everyone gets lost trying to figure out what's going on with your system
↧
FreePBX 13 NO GUI Web Access
↧
FreePBX 13 NO GUI Web Access
after a failed freepbx 13 upgrade.. AMPORTAL will not run. Below is the returned msg
# amportal
Fetching FreePBX settings with gen_amp_conf.php..
/usr/local/sbin/amportal: line 52: Whoops\Exception\ErrorException:: command not found
/var/lib/asterisk/bin/freepbx_engine: line 100: Whoops\Exception\ErrorException:: command not found
**** WARNING: ERROR IN CONFIGURATION ****
astrundir in '/etc/asterisk' is set to but the directory
does not exists. Attempting to create it with: 'mkdir -p '
mkdir: missing operand
Try `mkdir --help' for more information.
**** ERROR: COULD NOT CREATE ****
Attempt to execute 'mkdir -p ' failed with an exit code of 1
You must create this directory and the try again.↧
↧
Amportal returns error
2 posts were merged into an existing topic: FreePBX 13 NO GUI Web Access
↧
Amportal returns error
↧
Cant receive call
Hello,
I have big problem,
i setup trunk, inbound and outbound.
Right now, i can call using softphone to my cellphone (outbound)
but when i try to call my softphone (voip phone) from my cellphone, i got "this number is not in service"
i tried everything, and i everything seem to be set up perfect.
Here is my log file
[2015-09-14 15:51:15] VERBOSE[11231][C-00000013] netsock2.c: == Using SIP RTP TOS bits 184
[2015-09-14 15:51:15] VERBOSE[11231][C-00000013] netsock2.c: == Using SIP RTP CoS mark 5
[2015-09-14 15:51:15] VERBOSE[23099][C-00000013] pbx.c: -- Executing [18198015077@from-sip-external:1] NoOp("SIP/sip.us1.twilio.com-00000012", "Received incoming SIP connection from unknown peer to 18198015077") in new stack
[2015-09-14 15:51:15] VERBOSE[23099][C-00000013] pbx.c: -- Executing [18198015077@from-sip-external:2] Set("SIP/sip.us1.twilio.com-00000012", "DID=18198015077") in new stack
[2015-09-14 15:51:15] VERBOSE[23099][C-00000013] pbx.c: -- Executing [18198015077@from-sip-external:3] Goto("SIP/sip.us1.twilio.com-00000012", "s,1") in new stack
[2015-09-14 15:51:16] VERBOSE[23099][C-00000013] pbx.c: -- Goto (from-sip-external,s,1)
[2015-09-14 15:51:16] VERBOSE[23099][C-00000013] pbx.c: -- Executing [s@from-sip-external:1] GotoIf("SIP/sip.us1.twilio.com-00000012", "1?checklang:noanonymous") in new stack
[2015-09-14 15:51:16] VERBOSE[23099][C-00000013] pbx.c: -- Goto (from-sip-external,s,2)
[2015-09-14 15:51:16] VERBOSE[23099][C-00000013] pbx.c: -- Executing [s@from-sip-external:2] GotoIf("SIP/sip.us1.twilio.com-00000012", "0?setlanguage:from-trunk,18198015077,1") in new stack
[2015-09-14 15:51:16] VERBOSE[23099][C-00000013] pbx.c: -- Goto (from-trunk,18198015077,1)
[2015-09-14 15:51:16] VERBOSE[23099][C-00000013] pbx.c: -- Executing [18198015077@from-trunk:1] Set("SIP/sip.us1.twilio.com-00000012", "__FROM_DID=18198015077") in new stack
[2015-09-14 15:51:16] VERBOSE[23099][C-00000013] pbx.c: -- Executing [18198015077@from-trunk:2] NoOp("SIP/sip.us1.twilio.com-00000012", "Received an unknown call with DID set to 18198015077") in new stack
[2015-09-14 15:51:16] VERBOSE[23099][C-00000013] pbx.c: -- Executing [18198015077@from-trunk:3] Goto("SIP/sip.us1.twilio.com-00000012", "s,a2") in new stack
[2015-09-14 15:51:16] VERBOSE[23099][C-00000013] pbx.c: -- Goto (from-trunk,s,2)
[2015-09-14 15:51:16] VERBOSE[23099][C-00000013] pbx.c: -- Executing [s@from-trunk:2] Answer("SIP/sip.us1.twilio.com-00000012", "") in new stack
[2015-09-14 15:51:16] VERBOSE[23099][C-00000013] pbx.c: -- Executing [s@from-trunk:3] Log("SIP/sip.us1.twilio.com-00000012", "WARNING,Friendly Scanner from 54.172.60.3") in new stack
[2015-09-14 15:51:16] WARNING[23099][C-00000013] Ext. s: Friendly Scanner from 54.172.60.3
[2015-09-14 15:51:16] VERBOSE[23099][C-00000013] pbx.c: -- Executing [s@from-trunk:4] Wait("SIP/sip.us1.twilio.com-00000012", "2") in new stack
[2015-09-14 15:51:18] VERBOSE[23099][C-00000013] pbx.c: -- Executing [s@from-trunk:5] Playback("SIP/sip.us1.twilio.com-00000012", "ss-noservice") in new stack
[2015-09-14 15:51:18] VERBOSE[23099][C-00000013] file.c: -- <SIP/sip.us1.twilio.com-00000012> Playing 'ss-noservice.ulaw' (language 'en')Can anyone help me to fix this?
thanks alot
↧
↧
SIPstation trunks and RTP forwarding
We are thinking of getting a few Sipstation trunks.
We can source-restrict SIP 5060 to a SIPstation IP address or domain name, so that's good.
We don't feel comfortable opening RTP ports to the internet at large though . Although there is no threat of hacking into our system, Dos attacks would still be possible.
What is your take/experience on this, and how does Asterisk deal with a ton of unauthorized RTP traffic?
↧
SIPstation trunks and RTP forwarding
For RTP traffic Asterisk doesn't bind to 10000-2000 ports on startup. It binds to them dynamically. If you telnet to an asterisk system on UDP port 10000 and there are no calls happening you will just timeout.
↧
Licensing Log
If you send me a PM with your deployment ID I can double check and make sure that all your licenses are in line. (optionally you can open a sales ticket at support.schmoozecom.com to have someone take a peek.)
↧
FreePBX 13 Release Candidate
Please file a bug report for the above issues at http://issues.freepbx.org
↧
↧
FreePBX 13 Release Candidate
Creates a bug for the first issue so I could commit a fix
↧
SBC as an admission controller for remote extensions
Thanks dicko! Is there anything that needs to be changed with regards to fail2ban to make sure that it is able to stop any brute force attempts against a random port (as your ex., against 54667)? Also, I did some testing; asterisk will take a random port even if that's not it's default port. Do you still recommend the default listening port be changed from 5060?
↧
FreePBX 13 NO GUI Web Access
./fwconsole
Whoops\Exception\ErrorException: trim() expects parameter 1 to be string, array given in file /var/www/html/admin/libraries/modulefunctions.class.php on line 2128
Stack trace:
1. Whoops\Exception\ErrorException->() /var/www/html/admin/libraries/modulefunctions.class.php:2128
2. Whoops\Run->handleError() :0
3. trim() /var/www/html/admin/libraries/modulefunctions.class.php:2128
4. module_functions->_readxml() /var/www/html/admin/libraries/modulefunctions.class.php:622
5. module_functions->getinfo() /var/www/html/admin/libraries/utility.functions.php:1347
6. bootstrapparse_hooks() /var/www/html/admin/libraries/utility.functions.php:1322
7. bootstrap_include_hooks() /var/www/html/admin/bootstrap.php:257
8. require_once() /etc/freepbx.conf:9
9. include_once() /var/lib/asterisk/bin/fwconsole:12
↧
FreePBX 13 Release Candidate
Sorry, I'm new to the bug reporting thing. I'll use that form. Thx.
↧
↧
FreePBX 13 Release Candidate
Will the manual install on Ubuntu 14.04 support commercial modules eventually?
↧
Solved : Asterisk switched to 13, Digium phones module broken
This solution helped me. However, I had to use:
mv /usr/lib/asterisk/modules/res* /usr/lib64/asterisk/modules/
↧
Realtime CDR records
No per-se "server settings", if you want to use another mysql client you will need to set up a user/password that has the ACL to get to mysql. If you have an ssh client you can
ssh user@machine -p portyouuse "mysql -u youruser -pyourpassword -D asteriskcdrdb -e 'select * from cdr'"
You might want to set up a nopassword login using ssl keys for simplicity.
↧
SBC as an admission controller for remote extensions
fail2ban is as effective as the regexes you build to use against the log files that expose the IP addresses of the bad guys.
I will disagree with "asterisk will take a random port" , it won't. If you are talking about rtp connections, don't worry, apart from a vague possibility that a local host on your network might listen to a phone call without using srtp then there is no risk at all..
Yes 99.99% of all attacks originate on ports 5000-5999, 99.786% on 5060/5061, just don't unnecessarily expose your self .
↧
↧
FreePBX 13 NO GUI Web Access
There is a module on your system (NOTE: Nothing from Sangoma, a third party module). That has multiple < description > tags in it. This breaks all sorts of things of course (and is not to spec)
You'll have to find that module. Look in /var/www/html/admin/modules/*/module.xml
There will be a lot of them.
↧
FreePBX 13 Release Candidate
We have nothing on the roadmap for this at this time. Only CentOS is supported.
↧
OpenVPN Server - FPBX 13 RC1
This was from a fresh / stock / standard install from FBPX Distro so unless that module is installed by default, I didn't install or enable any additional modules.
↧