This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.
Presence Watcher BLF key? (DND vs Busy?)
Issues with logs and disk space
Any idea? please?
Usb install
did you get this solved?
Internal-only basic configuration for analog phones to call each other when running through Grandstream HT704
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
Issues with logs and disk space
the “drop” directory /etc/logrotate.d/ also contains pertinent rotation rules.
If you don’t have one that rotates the fail2ban logs then that is likely your problem, but you should investigate those log files before you rotate them, they will likely expose your real problem.
What can I use in a context to get the extension Outbound CID
Obviously. Or I wouldn’t set it up.
Who said anything about an outbound call. In fact, it is very clearing an internal call.
Multi SIP trunk problem with inbound calls from same FreePbx
Not obviously. You haven’t produced a sip debug to show that it is not working or what is actually being sent to the provider. So you will need to make another test call and this time instead of “asterisk -rvvvvvvvv” you just do “asterisk -r” followed by “sip set debug on” and make the call.
What can I use in a context to get the extension Outbound CID
Then you have all the needed info apart from the Warning about ‘my-operator,s,1’ you posted
Clean install, SIP connections 403 Forbidden
That is not correct and shouldn’t even be in there. Remove it.
Sip trunk getting poor voice quality when high traffic
im am new on Wireshark analysis, may I know how to you saw the call stop coming in at the same time?
Freepbx 14 default root password
the same happened to me after installation could not login with root default password,
but l reset the root password and managed to login.
Multi tenant simultaneous calls limiting
Hi,
I’ve got multiple clients running off the same cloud hosted fpbx 13 box, the calls to their did’s are all coming in on the same trunk, so I don’t want to limit calls at the trunk level, but at the route level or similar.
For example:
customer A has 5 phones, but is only paying me for 3 lines, I want to limit their simultaneous calls to 3 (across all 5 of their phones).
customer B has 10 phones, but is paying me for 6 lines, I want to limit their simultaneous calls to 6 across their 10 phones).
What’s the best way to achieve this?
queues?
in/out route?
is there a commercial module for this?
Thanks
Multi tenant simultaneous calls limiting
No, there are no commercial modules for this. FreePBX is not design for multi-tenant deployment. While you can use it as a base, you are going to have to customize the hell out of it for what you are looking to do.
Multi tenant simultaneous calls limiting
No, basically a multitenented Asterisk is doomed to ultimate f ailure
Multi tenant simultaneous calls limiting
I run Asterisk in a multi-tenant setup just fine. It just can’t be be the only thing in the mix.
Multi tenant simultaneous calls limiting
Tom & Dicko any recommended systems for multi tennant? Freeswitch? FusionPBX? other?
Multi tenant simultaneous calls limiting
I run multi-tennant too, I guess the only way to do this simultaneous call limiting is at the trunk level?
Sip trunk getting poor voice quality when high traffic
Packets 8745, 8746, 8748 and 8749 are from four different calls (port numbers are different).
Packets 8753, 8754, 8756 and 8757 are the next packets from the same four calls. The interval between packets for the same call is ~20 milliseconds, as expected.
After that, there is no incoming traffic for more than one second. However, traffic (that was received from extensions) continues to flow out to the trunks. This indicates (with reasonable certainty) that the the LAN is working properly and the PBX is not overloaded.
Based on the way traffic resumed, I feel that a provider problem is likely, though it’s conceivable that your router, ISP or an intermediate carrier buffered the four streams separately. If a test with your alternate provider shows that their RTP stops flowing at the same time, we can investigate the network issue further.
BTW, when you reach TheTestCall (by whatever means), when you hear the first “record your test message then press pound”, press #, then when you hear the main menu, press 4 for hold music. The music plays for about 10 minutes, which should be long enough to complete your test.
Ring Group 'Alert Info' for distinctive polycom ring
Just wanted to say thanks for this - this makes it so much easier to setup!
AT&T/Cricket Blocking Traffic
This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.