It’s nice that we now have the “config files!” people collaborating with the “web GUI!” people here on the same forum. 
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I finally gave up chan_sip for chan_pjsip and it was fine
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I finally gave up chan_sip for chan_pjsip and it was fine
I just say things. Sometimes people listen, sometimes people do things.
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Amazon AWS Chime Voice Connector Configuration
This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.
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Replacing +31 in outgoing calls
This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.
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Free pbx cti solution
i know Xtelsio and we are using it for out project with Asterisk
can we use it with Free PBX too?
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Error message on Apply Config after 10 mins of locked browser
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Playing opus records in CDR Reports
Submit a feature request.
Lots of the system is Open Source, so you might want to look at /var/www/html/admin/libraries/media/Media/Media.php and see if you can figure out how to add the transcoder.
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Extension unreachable / reachable
This is a new one for me. Do you have any idea what is requesting the extension be deleted?
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Module CustomContext
What are you actually trying to do?
You can’t make changed to the GUI through context changes, so if that’s where you’re starting, you need to look at GitHub for information on creating a new module.
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Cisco PAP2 with EndPoint Manager
Thankfully mine all ship in DHCP. Its a pretty basic, and easy config for myself. After 3 years for doing these it takes 30 seconds to do one, but unfortunately its time to move on from just me and have more functionality.
As for manually setting the rule, I have done that, except on port 83 as PBX default set it to that and I can’t change it. I have also attached a picture of the provisioning screen. Maybe I’m missing more, or maybe it’s just a cisco thing. I have never played with auto provision before.
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Extension unreachable / reachable
The extension isn’t being deleted. The registered contact to the AOR is being deleted. This is classic NAT/networking issues.
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Extension unreachable / reachable
It happens as a result of the device itself requesting the registration to be removed.
I’d suggest providing a SIP trace (pjsip set logger on) to show the actual traffic going on. As well - what is the device that is registering to this extension?
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One way Audio
The unhelpful (but correct) answer is that your NAT settings are incorrect.
The reason it’s unhelpful is because there are about five places that affect the NAT settings.
You need to look in all of the following:
- Under Advanced Settings -> SIP Settings, make sure all of the local nets and external addresses are set correctly.
- In your Trunk definitions, make sure any NAT settings you are using are correct.
- In your Extensions definitions, make sure the NAT settings are correct.
- If you have NAT settings in your actual phones (real or soft), make sure they are all set correctly.
- Make sure your Router isn’t helping you with SIP-ALG or some other impediment to success.
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One way Audio
Would this be the local IP of the remote phone?
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One way Audio
No, not at all i cannot even ping this IP address.
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Free pbx cti solution
FreePBX uses Asterisk, so the short is should be “Yes”. There may be some adjustments you’ll need to undertake (manual changes to /etc/asterisk/extenions.conf need to be moved to /etc/asterisk/extentions_custom.conf).
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Call Forward Ring Time
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One way Audio
So what are the subnets at the remote sites? You don’t have a VPN on every phone so they are going through your VPN at the router. That means they still have IPs from their local network. What is the IP of the phone that you are testing with?
Random IPs don’t just get shoved in there. Something is doing it.
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Restrict inbound calls to an extension - used for Door Lock Release
Current PBX Version:14.0.13.4
Current System Version:12.7.6-1904-1.sng7
Installing a Viking analog 3 relay module on a Vega analog FXS port. This will provide multiple door lock release for the Purchasing and HR departments.
I need to be able to restrict access to this extension to only allow calls from certain extensions.
What would be best practice?
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Extension unreachable / reachable
Hello,
There is the result of : pjsip set logger on
<— Transmitting SIP request (417 bytes) to UDP:172.16.2.4:5060 —>
OPTIONS sip:12@172.16.2.4:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.124.11:5060;rport;branch=z9hG4bKPj35ab618c-b431-4bd6-b610-365cd955985b
From: sip:12@192.168.124.11;tag=ed185835-4d62-4335-a638-5dc3c88031f4
To: sip:12@172.16.2.4
Contact: sip:12@192.168.124.11:5060
Call-ID: 2fdaf8d2-143d-435c-821c-b0d12aa4963f
CSeq: 65178 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-14.0.8.4(13.26.0)
Content-Length: 0
<— Received SIP response (486 bytes) from UDP:172.16.2.4:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.124.11:5060;rport=5060;branch=z9hG4bKPj35ab618c-b431-4bd6-b610-365cd955985b
From: sip:12@192.168.124.11;tag=ed185835-4d62-4335-a638-5dc3c88031f4
To: sip:12@172.16.2.4;tag=1755428817
Call-ID: 2fdaf8d2-143d-435c-821c-b0d12aa4963f
CSeq: 65178 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP2170 1.0.9.69
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
The device is a grandstream gxp2170
Thanks for your reply
Luca
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