↧
"Distro" in the cloud?
↧
Toll Free not working is it me or provider
That was my initial impression too, I am just at a loss as to the reason I cannot for the life of me receive calls on that toll free number. If I receive calls from others on the same provider it should process them the same way.
I will be bald before this all over!
↧
↧
Router port forwarding necessary
I’m getting conflicting information regarding port forwarding. Our NAT-enabled FreePBX box is sitting behind a Cisco router. Do we need to forward TCP 5060 and UDP 10000-20000 to the PBX? We do not have any external phones, they are all on the same LAN as the PBX.
I’m asking because we cannot receive inbound calls and when we make an outbound call, outgoing audio works, but we do not hear anything.
I thought that since our PBX establishes the outgoing connection to our SIP line provider we didn’t need to forward anything. Maybe I need to add something to our inbound ACL?
↧
Music on hold using soundcard line input
Hi Guys. I need help!
I’ve installed Alsa tools and utils with the intent of inputting audio on a soundcard for music on hold. Before we get into a discussion as to the legalities, I should mention that this is for a radio station installation of Freepbx.
I created a custom category for default MOH with line command “arecord -c 1 -f MU_LAW -r 8000 -t raw”
I can verify that sound card and alsamixer interface is working in CentOS. But I hear no audio on hold and I see this constant error in the log files
“[2018-10-25 20:26:36] WARNING[10781] res_musiconhold.c: poll() failed: Interrupted system call”
???
↧
White dashboard
Looks like your Apache setup is messed up - PHP probably isn’t working right.
↧
↧
Toll Free not working is it me or provider
This is a 100% you. Twilio sends incoming calls from at least 5 different IPs. Calls to your numbers could come from any of those IPs (it will be in the website FAQs/Docs). Chan_SIP only supports 1 IP per trunk, so you would need a Chan_SIP trunk for each of the IPs Twilio could send calls from.
↧
Callback feature ringing the number not more that 7 seconds
Logs for the 8 seconds after the callback is initiated would be really helpful. You can get them from /var/log/asterisk/full
↧
Music on hold using soundcard line input
That error occurs when the music on hold stream has no audio or is interrupted.
↧
E911 Notification - Dashboard widget / Email / SMS text / Paging
Have you tried escaping the pipe character?
↧
↧
E911 Notification - Dashboard widget / Email / SMS text / Paging
I did try escaping it… I"m going to try setting the command with a SET then executing it…
↧
E911 Notification - Dashboard widget / Email / SMS text / Paging
WOW… even if I SET the variable, then SHELL(variable), it STILL gives the same error.
↧
E911 Notification - Dashboard widget / Email / SMS text / Paging
ONE FINAL THING I could try doing… passing ${AMPUSER} and ${AMPUSERCID} as parameters, then have a bash script that accepts those parameters, and sends the mail.
What a complete PITA though…
At least then, it’s just calling a shell script and passing parameters to it.
↧
Can not connect to asterisk
so I got off the phone Sangoma and they explained it was all the google voice issue… cause I tired to get into it. so I removed it and it worked. thank you everyone for tryin to help me out
↧
↧
Router port forwarding necessary
If your system is registering with a remote ITSP, you need to whitelist the ITSP in all of your firewalls (including the one in FreePBX). The registration should set up the bidirectional path you need to get calls through the external router and back to the system. If there are problems getting this done, you should troubleshoot the connect using SIP DEBUG command (which vary depending on your channel driver). This will highlight if your sending commands to the right place and if the respondent is sending them back to the right place.
As long as you whitelist the service provider and blacklist everyone else, setting up the port 5060 connection on your edge router to forward to your PBX is good belt-and-suspenders approach and can make setup a little easier. It also solves problems where the remote system doesn’t connect back through the NAT correctly. Forwarding 10000-20000 on your external router to the PBX is actually safe - the large the range, the harder it is to spoof connections and trace calls.
If, on the other hand, you are using IP authentication (where your ITSP only knows your external IP address) then you will need to create the tunnel ports back to the server through the router.
Since you are registering, you shouldn’t need to set up the port forwards. The fact that they aren’t working may actually indicate that something else in your registration setup is incorrect. If you want to paste in the 30-or-so lines in the log where you registration is failing, we can help you troubleshoot that.
↧
Toll Free not working is it me or provider
This is coming from the “from-sip-external” context, so that’s an incoming (not outgoing) call.
Something in your setup is seriously horked. Time to start at the beginning.
Please anonymize and post your trunk config and outbound route config so we can look at them.
Also, how are these calls getting to the PBX? Normally, this would happen is you were connecting from another PBX and they were trying to call out to the wide world. Something definitely doesn’t add up. The clues point to a single, serious mistake. We should be able to spot it pretty quickly.
↧
Music on hold using soundcard line input
Where’s the rest of the command?
↧
Call recording not working after update
It seems that the issue took longer after changing settings, but it is still deleting call recordings with “Remove zero duration files” enabled even when they are greater than zero in duration.
For the past hour and a half, I kept the feature disabled and the recordings are kept. I think there might be a bug in how it’s handling the deletion of zero duration call recordings in the latest version for FreePBX.
↧
↧
White dashboard
(post withdrawn by author, will be automatically deleted in 24 hours unless flagged)
↧
E911 Notification - Dashboard widget / Email / SMS text / Paging
One last suggestion:
Wrap the echo command in quotes and escape the quotes inside the system() function.
exten => s,1,ExecIf($["${OUTNUM}"=“xxxxx”]?System(“echo \“Ext ${AMPUSER} - ${AMPUSERCIDNAME} just placed a 911 emergency call on ${STRFTIME(%C%m%d%y%H%M)}\” | mailx -r \“asterisk@pbx.mydomain.com\” -s \“911 Alert from ext ${AMPUSER} - ${AMPUSERCIDNAME}\” xxxxx@vtext.com”))
↧
Call recording not working after update
If you can repeatably make it fail, submit it on an Issues ticket and be sure to explain how you make if work and how you make it fail. A fix should come back fairly quickly.
↧