I suspect it had something to do with your Debian instance and the way it was installed. I haven’t seen any issues raised with FreePBX relating to your issues, though you certainly could - Issues · FreePBX/issue-tracker (github.com)
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Freepbx 17 problems feedback
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Problem with outbound call
Please confirm: Calling some landline numbers in São Paulo works, but others fail, even if they are served by the same carrier.
Unrelated to the failing calls, you have a problem where extension 223 is misconfigured and repeatedly trying to SUBSCRIBE to messages at address “1”, which Asterisk does not have, resulting in the log filling with garbage.
It’s been more than 10 years since I last saw a Poly SPIP and I don’t remember the details.
On a VVX, the relevant settings are
msg.mwi.1.callBack="*97"
msg.mwi.1.callBackMode="contact"
msg.mwi.1.subscribe=""
I’m guessing that you have something like
msg.mwi.1.subscribe="1"
and if changing “1” to “” doesn’t work, you need to find out what should go in place of the 1.
If you can’t easily fix this, just paste at least 30 seconds of log after the call was initiated.
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No audio on some PJSIP endpoints
We have a system that is having a weird issue with PJSIP extensions.
The issue is a lack of audio on PJSIP extensions on internal calls when connected from some public IP addresses. It started on 5/21/24 with no known changes to the PBX or ISP. The PBX has a public IP address and is one of many within the same data center and is the only system having an issue.
Here is what hours of testing determined. The problem is only present on PJSIP extensions connecting from some public IP addresses. In other words, calls between 2 PJSIP extensions in location A will work fine but between A and B there is no audio but the call will complete. If either is a CHANSIP extension there is no problem. The issue on exists on internal calls. External, using a trunk, are fine. If I enable call recording on an extension the audio is present on the call. I restarted Asterisk and rebooted the server with no affect.
The only way to get the audio to work was to disable direct media on the affected extensions. The issue seems to be related to the public IP address of the endpoint. When looking at packet captures there was no RTP stream or the port was in the 3xxx range.
This system was installed 3 years ago and this just started.
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No audio on some PJSIP endpoints
If the affected extensions are connecting via T-Mobile home or business internet, they made a gateway change two days ago, which might be your issue. See
Audio is present on the recording (but the parties still can’t hear each other), or does recording cause the call to start working? If the latter, try setting Direct Media for the extensions to No.
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No audio on some PJSIP endpoints
The affected ISP’s are at least Verizon Wireless and Altice. Frontier works fine. I suspect it is something within an external route.
Yes, the recording also has the audio. When recording is enabled there are no problems and everything works perfectly.
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No audio on some PJSIP endpoints
If Direct Media for the extensions was set to Yes, try setting to No and test.
If it was already No, or setting No doesn’t help, report whether enabling recording causes different codec selection or other media flow changes (encryption, packetization, etc.)
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No audio on some PJSIP endpoints
I had to turn off Direct Media to fix this.
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No audio on some PJSIP endpoints
OK, so there are (at least) three possibilities:
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A change in Asterisk or pjsip resulted in a new bug, causing direct media to be inappropriately invoked. I’m guessing that the LAN subnets at both locations (A and B) are the same (for example 192.168.1.0/24), but Asterisk is no longer taking into account that direct media won’t work if the public IPs are different.
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The bug has been there all along, but a change at one of the locations, e.g., new ISP, new router, firmware upgrade caused the LAN subnets to match.
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Both A and B have a SIP ALG enabled, which pretends (to Asterisk) that the endpoint is on a public IP. Asterisk assumes that the two public IPs can communicate directly and enables direct media. In this case, it’s necessary to manually turn it off.
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Freepbx 17 problems feedback
I just installed Freepbx 17 Beta on Debian 12 only had one small issue with th script but I took care of it by installing these 2 things as per errors in the Logs:
fwconsole ma install vqplus
fwconsole ma enable vqplus
Script worked perfectly after that, and my install works perfectly. As for the Debian CD/DVD repo, Debian does this by default from a Vannilla Install of Debian 12 from DVD One can just comment it out and use online repos.
@adell4444 Do you know if I keep updating this instance, can I just keep it and update it until FreePBX is stable and ready for production? Do you know if this version can be used for that?
One last question does anyone know how far the Commercial modules are away from being ported to Debian? My most important one needed is FaxPRO.
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Sangoma Talk disabling extension automatically
It’s a pretty simple setup: 2 extensions, 2 licneses. The first one is enabled and working fine, the other when I enable it and send the invite it then tells them that they’re logged out and when I look at the module it is no longer enabled for that extension. Asterisk sees a registered device though, even though the app clearly isnt working.
We had some problems with permissions on this iphone and i’ve sent the email a number of times to a few different email addresses. Dont know if that is affecting anything.
What am I missing?
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Sangoma Talk disabling extension automatically
+1 for weird sangoma talk issues. Mine likes to randomly disable extensions.
Some things to try:
- Disable sangoma talk and re-enable the service.
- Try to DISABLE it again (even if it’s already disabled) Submit, Reload. Enable, submit, reload.
- Check your version of the Sangoma Talk Module and Sangoma Realtime API. Update.
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Smart BLF - P Series XML
Thanks for the heads up on needing the P330. I would have been mighty upset (that I didn’t read) that it’s not compatible with a P325.
And hopefully it is that straight forward… Will update this thread if and or when I upgrade phones.
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Sangoma Phone desktop softphone headset support
Yeah it’s a bummer that there is no support for other big headset vendors like Plantronics or Jabra. Funny thing is when you look into the application directory of Sangome Phone Desktop you can find the Jabra Node.js SDK in the node_modules folder. So it looks like they had the plans to implement Jabra support. But honestly I have lost hope. Also we have sometimes the issue when you push on one of the buttons on our Jabra Evolve2 65 headset Sangoma Phone Desktop just suddenly starts ringing . The only thing to stop the ringing sound is to exit the application and restart it.
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No audio on some PJSIP endpoints
Speaking from an Asterisk perspective, that code hasn’t really been touched in… a long time. The “don’t do direct media if behind NAT” is a separate option in PJSIP disabled by default, and it looks at whether the IP address and port defined in the SDP is different than the source IP address and port of the media.
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No audio on some PJSIP endpoints
Regarding chan_sip, the OP may will have copy and paste coded the configuration and used canreinvite=no, without realising it is an obsolete name for directmedia=no. They may have thought they hadn’t set directmedia off, for chan_sip, when, in fact, they had.
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Freepbx 17 problems feedback
In the past, I don’t think you could update from beta straight to a stable release. With the Debian version I’m not entirely sure how it will work. I’m also not sure how far along they are with porting the commercial modules - @lgaetz any insight on either of these questions?
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FXS and FXO on the same T1?
Hi!
I have been running FreePBX 16 and I have a single port T1 card feeding a channel bank (adtran 750) with FXS cards feeding some analog devices on my system. I get my telephone service via SIP, which works fine, except when the internet fails. I have an ATA connected to a different service provider and connected to a different ISP which remains up when the primary ISP goes down. I have added an FXO card into the channel bank to use the analog ATA as a backup trunk when the main SIP provider is unreachable. The issue is, that the T1 configuration in DAHDI does not have an option to make some channels FXS and others FXO, it appears to be an “all or nothing” as far as what I want the T1 to be, all channels have to be FXO or FXS according to the configuration. Is there a way to set up, say, channels 1 thru 16 as FXS and 17 thru 20 as FXO? It’s been a looooong time since I set it up!
Thanks! I’m probably missing something silly but I thought I’d ask!
-Rick
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FXS and FXO on the same T1?
channels are assigned in /etc/dahdi/system.conf, channels and grouping in /etc/asterisk/chan_dahdi*.conf
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FXS and FXO on the same T1?
Those files have large warnings at the top saying that they need to be managed via GUI and if they are changed, the GUI will overwrite any changes I make.
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FXS and FXO on the same T1?
I have found the dahdi management module not too good unless you have a PRI, make the edits and disable/uninstall the module.
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