Maybe a log trace will highlight which piece is missing?
https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs-PartII
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Multicast Paging Phone Ring
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Attended transfer code now drops calls?
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Where Is Video Call Recording File
This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.
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Web interface port from command line?
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Bizarre Issue between S505 and S500
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Cisco 7975 and FreePBX
I think this might be a bug. I went into the trunk settings to check the rport setting, it was listed as on. So without changing anything I clicked save and re-applied the config, this did not fix it (didn’t think it would). Went into the Asterisk console and turned on pjsip logging. and could see the registration request being sent with no reply coming back.
So out of desperation I went back into the trunk settings, scrolled to rport setting, click “On” even though it was allready set to that, saved and applied the setting and the trunk immediately back online and is working.
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Zulu UC3 as IM/XMPP client
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Cisco 7975 and FreePBX
If there is a bug, please post it at issues.freepbx.org .
However, I don’t understand what you are saying:
Registration for what? AD does not accept registration. Do you mean that saving the trunk config caused your Cisco phone registration to fail? Or is this registration for some other extension or trunk?
In any case, Force rport should only apply to replies to incoming requests; outgoing REGISTER or OPTIONS should be unaffected. And, on my AD incoming trunks, the INVITE’s Via header has the rport tag, so ‘forcing’ it should have no effect at all.
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FreePBX 15.0.16.53 - commercial modules keep reinstalling
I agree. I checked my system again today. No commercial products have been reinstalled over the last week.
I hope that this stays resolved. I found it frustrating.
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Suspicious activity
Did it this morning. Thanks, I agree with your assessment.
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Cisco 7975 and FreePBX
Sorry, I am new to this, and still learning how it all works. I guess it’s not a registration, but a “SIP request” which received no response.
I just saw this over and over again:
<--- Transmitting SIP request (434 bytes) to UDP:50.22.101.14:5060 --->
OPTIONS sip:sbc.anveo.com:5060 SIP/2.0
Via: SIP/2.0/UDP {myPublicIP}:5060;rport;branch=z9hG4bKPj02db24b5-c1ba-4d1f-a71d-dd93ccbcff67
From: <sip:AnveoDirect@172.17.50.99>;tag=61b15994-cd13-420d-bdb4-13dddddc8a40
To: <sip:sbc.anveo.com>
Contact: <sip:AnveoDirect@{myPublicIP}:5060>
Call-ID: acc2ea00-e1ea-4b93-97d2-32e011b4a8d9
CSeq: 22357 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-15.0.16.53(16.9.0)
Content-Length: 0
With nothing coming back.
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Capture user input and pass it on to extension and VM
Thank you for your input.
I have successfully completed part 1 of this project…
I am now able to get user input, and replace the CID name with the input provided.
Next step is to route calls based on whatever they entered.
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Sangoma phone account will not register
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Cisco 7975 and FreePBX
Asterisk implements the Qualify function by sending periodic OPTIONS requests. It considers the trunk to be online if it gets any response, even an error.
Assuming that you still have the FreePBX firewall disabled, my guess is that your hardware router/firewall is blocking the replies. If it has a SIP ALG, try turning that off. Otherwise, post details and someone here might know what setting is responsible. If you have a way to capture traffic on its WAN interface, that would confirm or refute whether it is to blame.
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GS GXP2170 and End Point Manager
Thanks for the reply,
I did run the log tail and it seems the phone is not even hitting the server at all. I do have the firewall setup correct as i have other phones registered from that ip address on that system. So, yes the public ip is trusted zone.
I guess i need someone with grandstream experience on this.
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Cisco 7975 and FreePBX
It’s probably not my opnSense firewall, and yes the freePBX firewall is off (and will remain off). This setup has worked 100% for a few months now. As soon as I toggle the rport setting to off on the extension, replies from Anveo stop reaching asterisk. Even on the off chance this is firewall related, it is still very much unexpected behavor of the setting.
I have verified that toggling the rport setting to off on an extension causes the trunk to go offline, and then turning it back on causes the trunk to come back online.You have to turn rport off on the extension and apply the settings. The trunk will go offline. Then if go the trunk settings and find the rport setting and click “on” (even though it is shown as all ready on), save the trunk settings and then apply the config and the trunk will then become available again and everything works. This is repeatable, and no other settings were changed in the process, and this is why I suspect it to be a bug.
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Ring before Multicast Paging
I think this may be a Sangoma settings issue. I can get a Multicast paging to go out from FreePBX and the Sangoma (s206) will pick it up. However, it makes a ring before it goes to the page. I’m wanting it to not ring first and just open up to the page. Can anyone confirm if this is a standard behavior with Sangoma phones and Multicast Paging from FreePBX?
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Broke my out going calls, All Circuits Busy
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Call Confirm Issues?
I have just tested the extension level Call Confirm and that does work.
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Mail Queue - messages not going to email anymore
Thank you! I got it all fixed. Two online forms filled out, one to Comcast and the other to Vade Secure Abuse contact and 3 phone calls to Comcast. 1 week of down time.
Comcast told me they would call back - never did. And I never got an email back from Vade. Comcast emailed back with and said I wasn’t blacklisted. That was before I posted. Hope it might help someone else.
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