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Call rejected because extension not found in context 'from-internal'
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Call rejected because extension not found in context 'from-internal'
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Vega - Blocked CallerID Received as Anonymous Inbound SIP Call
Looks like you’re using a PJSIP trunk to the Vega. See Asterisk SIP Settings, PJSIP tab and check the Endpoint Identifier Order. Assuming your Vega has a fixed ip, you want IP to come before Anonymous.
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Google Voice Motif
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Vega - Blocked CallerID Received as Anonymous Inbound SIP Call
Thanks for the pointer, Lorne. Looks like we already have it set that way, though.
Could this be related to the “Local Domain” Setting on the Vega in the SIP profile? I have noticed that it is set to the IP of the PBX, and not the IP of the Vega, and that just seems wrong to me. You can see that the “P-Preferred-Identity” and ther “From:” in the normal call both use the x.x.x.107 (PBX) IP address, not the x.x.x.108 (Vega) IP address.
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Vega - Blocked CallerID Received as Anonymous Inbound SIP Call
Try reversing the positions of IP and Username.
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Vega - Blocked CallerID Received as Anonymous Inbound SIP Call
Thanks again. Gave that a go, but no change. A call with blocked CallerID still displays the "Received incoming SIP connection from unknown peer..." message, while a normal call does not. Similarly, the log messages for the anonymous call look like PJSIP/anonymous-00012d74 instead of PJSIP/VegaPJSIP-00012d70.
I don’t think that the username/IP setting will make a difference (but then again I’m far from an expert), as the call is presented as “anonymous@anonymous.invalid”?
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Vega - Blocked CallerID Received as Anonymous Inbound SIP Call
Some of these changes require restarting (not just reloading) Asterisk.
If you still have trouble after making IP highest priority, try putting the IP address of the Vega in the Match (Permit) field (and restarting Asterisk).
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Vega - Blocked CallerID Received as Anonymous Inbound SIP Call
Thanks to you @lgaetz and @Stewart1! Changing the order and then executing fwconsole restart seems to have resolved this!
Call me crazy here, but doesn’t it seem like undesirable behavior for the Vega to insert “@anonymous.invalid” into the From: header? Seems like we do know where the call is coming from, just not the user/number, so Anonymous@192.168.123.108 would be more accurate and simultaneously avoid weird problems like this, no?
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Vega - Blocked CallerID Received as Anonymous Inbound SIP Call
You’re crazy 
The from header is usually set to the identity of the caller, unless you deliberately set it to some other value and use different SIP header(s) to encode the caller’s CID. In this case, it’s not the fact that the From header was set to anonymous that caused the failure, it was the fact that there exists a PJSIP endpoint called ‘anonymous’ generated by FreePBX behind the scenes that was matching the incoming INVITE. The same problem would have occurred if you had a PJSIP extension numbered 90210 and you were receiving calls with a malformed CID of 90210, the inbound call would have matched the extension and would’ve been misrouted as a result.
I assume you have PJSIP extensions, make sure that the change has not negatively affected outbound calls from any PJSIP extensions.
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Vega - Blocked CallerID Received as Anonymous Inbound SIP Call
Thanks, @lgaetz. That does clear it up some, though it still seems weird to me that the hostname would be affected by what the callerID is. 6035551212@192.168.123.108 for a normal call and anonymous@192.168.123.108 for a blocked call would make a lot more sense to me. Of course, your explanation above means it would have resulted in the same behavior, but still!
Also, it’s possible to modify the user for blocked calls to something other than “anonymous” using the Vega SIP advanced settings, so perhaps it would make sense for the FreePBX Vega setup tool to change this to “Blocked” or something else that would not cause this issue? Thoughts?
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Query extension status and do something
Hi guys,
I’m not a huge expert in Asterisk, but love the solution…
How can I do a script that query an extension status (register or unregistered) an if it’s unregistered make a call to a xxx-xxx-xxxx number and play a recording.
Thanks,
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Query extension status and do something
(post withdrawn by author, will be automatically deleted in 24 hours unless flagged)
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Query extension status and do something
If your extension was ‘6904’ and your recording is ‘your-recording.wav’ in /var/lib/asterisk/sounds/(your-language)/custom , then a good start would be (from a shell)
[[ "$(rasterisk -x 'sip show peer 6904'|grep Status)" =~ UNKNOWN ]]&& echo -e "Channel: Local/xxxxxxxxxx@from-internal\nApplication: Playback\nData: custom/your-recording\n" > /tmp/call.file;mv /tmp/call.file /var/spool/asterisk/outgoing/
It doesn’t test for ‘Registered’ so much, just if it is ‘available’
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How to solve "Reload failed because retrieve_conf encountered an error: 1"
no sir i just download the updates
i thought it will fix the cronmanager issue
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How to solve "Reload failed because retrieve_conf encountered an error: 1"
Sir can you gave me an example. sorry for being noob. just new here
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[Fail2Ban] SIP: banned on localhost
Hi,
I mount a FreePBX 15 Distro Sangoma Linux release 7.5.1805
The PBX is sending Email
[Fail2Ban] SIP: banned 192.168.x.xxx on localhost
To none@yourpbx.com
Hi,
The IP 192.168.x.xxx has just been banned by Fail2Ban after
7 attempts against SIP on localhost.
Regards,
Fail2Ban
First question is where do I set the To:
in the System Admin Notification setting it is set to a valid email not to the none@yourpbx.com
Second question is where do I set the rang of IP that I want to let pass thru
Thanks
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[Fail2Ban] SIP: banned on localhost
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[Fail2Ban] SIP: banned on localhost
The email was set in the E-mail:
It is my valide email
in the Whitelist it was set to
127.0.0.1
192.168.0.0/16
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RTP ports and iptables
ok will do.
thanks everyone
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