Ok so I’ve chmod777 the recordings folder and it didn’t help. In the logs there is an error:
file.c: Unable to open file: path - No such file or directory
and then
app_mixmonitor.c: Cannot open: path
That’s all I can see.
@edit
So I’ve created a forest inside witch matches the forest in the error - there were date folders missing. Now I’m getting permission denied error.
You gotta give permission for the asterisk user.
I belive that I have to create user on ReadyNAS with name asterisk and same password as on the PBX server. I’ve chown’ed and chmod’ed the folder once more and I’m still getting permission denied error. I think it’s caused by the ReadyNAS server that doesn’t have asterisk user and doesn’t let him write down anything. What will be the password for asterisk account on PBX server? Is there a standard password or previous person that was managing it set it up and I don’t really have access to it now? In that case - can I change it somehow?
So you have two PRIs to the pstn.
Let’s say you have those connecting to PRI port 1 and 2 on the gateway.
Is your goal then to route all calls common from your 6xxx freepbx extensions out through PRI port 1, and calls coming from 7xxx through PRI port 2?
Those gateways are running Asterisk and you are dealing with a dual nic Asterisk situation.
I am using Digium PRI gateways myself, albeit the smaller models with two PRI ports and no dual nic.
I think you can have that easier.
Just run one trunk between the gateway and Freepbx.
You can route via dialed number on the gateway.
You e.g. you can say if a call comes into the Digium and the dialed number matches a specific string, then send it out port 1 or 2.
If you absolutely must send all calls from your 6xxx freepbx extensions out through port 1 and 7xxx through port 2, I would have only one trunk on Freepbx, but two outbound routes.
Force 6xxx to use outbound route 1 and pretend a unique identifier, an extra digit string to the dialed number, e.g. 22, which you remove again on the gateway, but which allows your gateway to route calls starting with 22 out a specific port, whereas others will be sent out port 2.
I can show you how to configure routing rules on the gateway to make this work.
I am pretty confident that the solution I described above is gonna work for you, although there might also be an option to route the call on the gateway depending on which nic it came in on.
Post a screenshot of a routing rule on the gateway, so we can look at the options you have there.
But when i am removed the (!) ,i am not able to start the asterisk.
You are replying to a 2 year old post.
Little things have changed since…
Please post the full error you get, if you can, please open a new topic as well…
Hi,
We had to “upgrade” to the commercial endpoint manager as the OSS was no longer getting regular updates.
I’m able to provision the phones OK, but they (Snom 720) ignore the custom dialplan, referenced within the basefile.
</phone-settings>
Reverting to the old files backed up from the OSS version, the phones load the dialplan file OK.
Brgds
Tom
Thanks for pointing me in a direction, it is appreciated
Dear Team,
When i am trying to install freepbx 14 in cent os 7 with 5.6 and mysql 5.7 i am getting the below error.
**[root@ip-172-31-31-231 freepbx]# ./install
Database engine [mysql]:
Database name [asterisk]:
CDR Database name [asteriskcdrdb]:
Database username [root]:
Database password:
File owner user [asterisk]:
File owner group [asterisk]:
Filesystem location from which FreePBX files will be served [/var/www/html]:
Filesystem location from which Asterisk configuration files will be served [/etc/asterisk]:
Filesystem location for Asterisk modules [/usr/lib64/asterisk/modules]:
Filesystem location for Asterisk lib files [/var/lib/asterisk]:
Filesystem location for Asterisk agi files [/var/lib/asterisk/agi-bin]:
Location of the Asterisk spool directory [/var/spool/asterisk]:
Location of the Asterisk run directory [/var/run/asterisk]:
Location of the Asterisk log files [/var/log/asterisk]:
Location of the FreePBX command line scripts [/var/lib/asterisk/bin]:
Location of the FreePBX (root) command line scripts [/usr/sbin]:
Location of the Apache cgi-bin executables [/var/www/cgi-bin]:
Directory for FreePBX html5 playback files [/var/lib/asterisk/playback]:
Assuming you are Database Root
Checking if SELinux is enabled…Its not (good)!
Reading /etc/asterisk/asterisk.conf…Done
Checking if Asterisk is running and we can talk to it as the ‘asterisk’ user…Yes. Determined Asterisk version to be: 14.7.6
Preliminary checks done. Starting FreePBX Installation
Checking if this is a new install…Yes (No /etc/freepbx.conf file detected)
Database Root installation checking credentials and permissions…Error!
Invalid Database Permissions. The error was: SQLSTATE[HY000] [2002] No such file or directory
can anyone help on this?
Create only one trunk and one endpoint on the gateway.
In FreePBX create another outbound route for all yor 6xxx extensions, with 6xxx in the CallerID field in the dial patterns tab, and 222 in the prepend field. Put that route on top of the order.
Now on the Digium, do this in your Routing rule. Send call through the port you want your 6xxx to go out on.
Have a second routing rule without the 222 and route through the other port.
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
Thanks for replying. I am using Chan_sip as trunk. Also this does not occur when dialing extension to extension that operates perfectly. I believe ports 80,81,83,85 and 84.
Here’s the relevant data from ‘full’ log having switched pjsip logging on and conducted the test call with wrong CID set (to mobile number 01701213434).
Two points suspicious to me:
Occurence of leading ‘0’ at dialed number ‘00692575632’. We do not use leading ‘0’ for external calls.
Section at lines 288-297. Authentication error on INVITE. At FreePBX Webinterface > Connectivity > Trunks > edit my trunk > pjsip Settings > General > Authentication we selected option Outbound. Perhaps the cause of the problem? But why?full.tgz (4,4 KB)
Hi.
For some reason my gateway GXW4108 does not connect the trunk. So line with outgoing call ability connected with extention. I guess that I need to add extensions_custom.conf rules for outgoing calls or there are pretty solution with freepbx web gui?
Doesn’t look like you have your SIP ports open. Figure out what ports you are using, and open it from your SIP Provider only.
(By default, PJSIP is 5060, ChanSIP 5160, you will also need to open 10000-20000)
HI All,
I have purchase Appointment Reminder i have configured I have issue is calling before the day start time the status is Not Running and still making calls is wakening the customers at 05:00 AM when it should not call at that time the start time is 09:00 till 17:00 M-F and Saturday and Sunday is from 11:00 till 17:00
Thank you
Is your system time set to the correct time zone?
I have a customer who wanted 2 of the same phones with the same extension at 2 different spots so I set them up with PJSIP using multiple contacts. They wanted some modifications made to the buttons on the phones so I did that and pushed out the update but it only updated 1/2 phones and they are the same models.
I looked at asterisk and it just lists one number for the extension and when I try to sync-yealink it then it still only updates one phone. How do I go about updating both phones using pjsip notifies?
