I also saw the line “Started music on hold” in the logs. Sorry for not attaching it.
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Music on Hold is played sometimes
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Music on Hold is played sometimes
I uploaded in .mp3 format and converted to all formats available to me - alaw, g722, gsm, ulaw.
Is your selected MOH audio playing at the FreePBX side ?
Well, if the caller’s call has been established with an operator, and call forwarding is required, then in this case the music plays with 100% probability.
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Music on Hold is played sometimes
Hmm. Thanks for the detailed answer, I’ll try to dig in this direction.
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Music on Hold is played sometimes
Replacing the ring group with a queue actually solved the problem, but I’m still wondering what the problem was.
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Sangoma S serie - TLS ciphers incompatibility
It’s unlikely to find firmware support for a phone that reached its end of life (EOL) eight years ago, especially since it was replaced by a newer model.
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FreePBX 17 install issues 2
I have raised these issues as bugs #128, #132. #134.
I did eventually find where to enable ARI but that just caused the message to change to The Asterisk REST Interface is not able to connect please check configuration in advanced settings.
After some more reading and guessing I solved this - see bug #132. I copy the relevant notes below:
- In Advanced settings: Set
Display Readonly SettingstoYesand also setOverride Readonly SettingstoYes- Look in settings
ARI UsernameandARI Passwordfor any non-alphanumeric characters and REMOVE. I found/+as part of the name.- Submit and Apply Config.
- Remember to turn off the settings made in step 1.
Problem solved.
So this is really an old bug - the system is still automatically generating random user/password values which are invalid.
Also, if it is important that by default ARI should be turned off, then wherever this results in a failure (like in Asterisk info which I would think is a core option) then clear instructions on how to turn it on should be included.)
PS: The Asterisk logfile is now called System logfile - an odd name change.
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Warnings after restore
After a restore I got these warnings :
npm WARN deprecated nomnom@1.8.1: Package no longer supported. Contact support@npmjs.com for more info.
npm WARN deprecated is-buffer@1.0.2: This version of ‘is-buffer’ is out-of-date. You must update to v1.1.6 or newer
Is it something to be worried about ?
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How to redirect an ip address to a domain name
how to redirect an ip address to a domain name on freepbx locally
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Configuring bind9 on FreePBX
how to redirect an ip address to a domain name on freepbx locally
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Module connection error - FastAGI failed
Hello.
After restarting Asterix, warnings appear in the console panel when running the -rvv command:
[2024-05-06 11:28:36] WARNING[2345][C-00000006]: res_agi.c:2038 handle_connection: Connecting to '127.0.0.1:4573' failed for url 'agi://127.0.0.1/dialparties.agi': Connection refused
[2024-05-06 11:28:36] WARNING[2345][C-00000006]: res_agi.c:2098 launch_netscript: Couldn't connect to any host. FastAGI failed.
[2024-05-06 11:28:50] WARNING[2272][C-00000005]: res_agi.c:2038 handle_connection: Connecting to '127.0.0.1:4573' failed for url 'agi://127.0.0.1/missedcallnotify.php': Connection refused
[2024-05-06 11:28:50] WARNING[2272][C-00000005]: res_agi.c:2098 launch_netscript: Couldn't connect to any host. FastAGI failed.
The station is normal working, what’s wrong?
*FreePBX 16.0.40.7 / Asterisk 19.8.1
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Sangoma S serie - TLS ciphers incompatibility
Ok thanks for your answers.
At least this thread documents the problem so futur people will understand why thoses phones cannot connect on newer HTTPS site/provisioning (it took me hours to find why it was not working).
I will try to have a look at the latest firmware, we are able to decompress it with “binwalk -e” but I have very little hope
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Configuring bind9 on FreePBX
What does this mean? The direction of translation is domain name to IP address. Whilst you can use PTR records to find out the domain name(s) for an IP address that doesn’t change where traffic goes.
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Polycom without microphone
Hi guys, here we use Polycons phone to make and receive our calls.
Friday the microphone of one of those stopped working on calls, if I test the hardware it records and reproduces my voice. But when I call someone they can’t hear me, but I can hear them.
I don’t know if this is some kind of configuration, or if the phone is dead.
Can you help me?
The model of the phone is a Polycom IP330 SoundPoint
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Sangoma S serie - TLS ciphers incompatibility
Once I have everything working, I never update, never ever. Update just breaks things that were once working.
I have both S500 and S505 deployed on a system that are currently using 3.0.4.88 (November 5, 2017)
TLS provisioning is working on that firmware version.
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Music on Hold is played sometimes
This is consistent with what I posted above. A queue will answer the calling channel immediately while waiting for an agent to answer, where a ring group does not.
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Music on Hold is played sometimes
This is not true. Straight from the docs:
This application does not automatically answer and should be preceeded by an application such as Answer(), Progress(), or Ringing().
Read more here:
https://docs.asterisk.org/Asterisk_20_Documentation/API_Documentation/Dialplan_Applications/Queue/
Edit: Here is the first few priorities of FreePBX generated dialplan for a queue
exten => 1000,1,Macro(user-callerid,)
exten => 1000,n,Set(__MCQUEUE=${EXTEN})
exten => 1000,n,Answer
So app_queue isn’t automatically answering the call, it’s being done by the Answer() application before the call hits the actual queue.
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PJSIP and 911 location
Searching trying to find the “most up to date” method of dealing with clients that have 1 extension number registering from 3 different physical office locations, using PHSIP. Need to pass the correct e911 info if 911 is dialed from any of the offices.
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Music on Hold is played sometimes
A key difference between a queue and a ring group, is that queues can ring out, for the caller, for many minutes, whereas ring groups are assumed to be answered quickly. Whilst, in some cases, this may not be the reality, if you didn’t answer the call early, for a normal queue, the call would be ended by the network, before it reached an agent.
If answer times are expected to be long, Queues are the construct you should use.
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Music on Hold is played sometimes
How fast a Ring Group can answer is a business logic decision. In FreePBX a ring group has a max ring time of 300 seconds. However, that is a limitation put on by FreePBX as all a Ring Group does is issue a Dial() command. That command could have one or more endpoints being dialled at the same time and much like any other Dial() command if no timeout is set the default is 136 years. A ring group does not hold the caller, it just dials the destinations as it is programmed to do. If none of the destinations answer, the call is routed to a post destination.
A queue on the other hand will hold the call for as long as told before even sending it to an agent. During that time the caller will hear hold music and could be given options to leave the queue. Much like a ring group, the queue will call the agents based on the strategy and how long the timeout is set for when calling agents. If none of the agents answer the call is pulled back into the queue. Now at that point the call can be held longer and the agents retried or the queue could kick the caller out and route it to a post destination.
I can program my queue to act just like a normal ring group but I can’t program my ring group to act like a queue.
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Polycom without microphone
It’s such an old phone. Have you tried making a call via speakerphone to see if that works instead of the handset?
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