That is a question for the FOP2 developer.
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FOP2 Avoid other agents pause
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Billing Module
Hey hey hey. Your experience in telecomms does NOT entitle you to post here personal attacks.
Kindly please edit your post and remove these personal attacks.
Thank you
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Hook file '/var/spool/asterisk/incron/firewall.stopfirewall' was not picked up by Incron after 5 seconds. Is it not running?
@lgaetz
[root@lyre ~]# service incrond status
Redirecting to /bin/systemctl status incrond.service
● incrond.service - Inotify System Scheduler
Loaded: loaded (/usr/lib/systemd/system/incrond.service; enabled; vendor preset: disabled)
Active: active (running) since Sat 2020-01-11 16:40:50 EST; 1 day 16h ago
Process: 19375 ExecStart=/usr/sbin/incrond (code=exited, status=0/SUCCESS)
Main PID: 19376 (incrond)
CGroup: /system.slice/incrond.service
└─19376 /usr/sbin/incrond
Jan 13 07:45:01 lyre.ad.snowpond.org incrond[19376]: (system::sysadmin) CMD (…
Jan 13 07:45:01 lyre.ad.snowpond.org incrond[19376]: (system::sysadmin) CMD (…
Jan 13 08:00:01 lyre.ad.snowpond.org incrond[19376]: (system::sysadmin) CMD (…
Jan 13 08:00:01 lyre.ad.snowpond.org incrond[19376]: (system::sysadmin) CMD (…
Jan 13 08:15:01 lyre.ad.snowpond.org incrond[19376]: (system::sysadmin) CMD (…
Jan 13 08:15:01 lyre.ad.snowpond.org incrond[19376]: (system::sysadmin) CMD (…
Jan 13 08:30:01 lyre.ad.snowpond.org incrond[19376]: (system::sysadmin) CMD (…
Jan 13 08:30:01 lyre.ad.snowpond.org incrond[19376]: (system::sysadmin) CMD (…
Jan 13 08:45:01 lyre.ad.snowpond.org incrond[19376]: (system::sysadmin) CMD (…
Jan 13 08:45:01 lyre.ad.snowpond.org incrond[19376]: (system::sysadmin) CMD (…
Hint: Some lines were ellipsized, use -l to show in full.
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BLF's not working right with Yealink 83.0.X firmware. Response from Yealink
Forgive my ignorance on patches/updates, but the Asterisk ticket says it’s for version 15. Would this patch be applied to 16 as well I’m assuming?
Thanks!
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BLF's not working right with Yealink 83.0.X firmware. Response from Yealink
The change goes into all applicable supported branches. That would be 13, 16, and 17.
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FOP2 Avoid other agents pause
IIRC you can prevent this in the template
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Caller ID Manipulation
I guess. I do this for a school district that has multiple schools and after school programs. People call in and select the school program from an IVR/directory, it uses FollowMe to send to the call, no confirmation send call to voicemail which emails attachment of voicemail to person that was called. They want the callers CallerID for callbacks and they seem to manage fine doing it this way.
Also nothing is to say they can’t force a fixed CallerID for the school’s CallerID so it looks like the school is calling them like you suggested just setting Call Confirm helps control the call. However, I wouldn’t bother messing with CallerID Name as the receiving carrier could just ignore and do their own dip on the name.
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Outside users calling outside the country? Fraud?
I got an email from my phone company about fraudulent calls. I don’t expose the system to the internet so they are dialing in and then dialing another number over the phone. I’ve had it before where they guessed a SIP extension password and made calls that way. But that isn’t the case here since I don’t forward any ports from the firewall. I don’t use DISA. Any thoughts by looking at the logs?
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Config.php hacked - need to replace
You can try to install another clean system and export what you may have and once this new install ready, then use it.
To be honest, the wrong is done and we don’t know what is hacked right now. So, IMHO, no need to get some risk.
Improve you security for your system appplying good rules in the firewall. (the best is , to accept only trusted ip address, and drop the others).
Otherwise, you can compare any settings manually and swap to the new system once ready.
Also, you can use fwconsole ma downloadinstall for all module one by one, and next, take a look your dialpan, sip settings…etc (extensions_custom.conf, sip ,pjsip, manager…etc) if some hacked contexts, manager or devices are present somewhere in your system.
rkhunter is useful to check if there’s some stuff wrong in your system. You can install it even if it’s a little bit old.
Don’t forget to update your system (O.S + Freepbx) as often as possible.
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Res_srtp.c: SRTCP unprotect failed because of authentication failure
As I recall, this happens when Asterisk tries to unencrypt and authenticate an incoming RTCP packet and there’s a problem in that process. It can either mean packet corruption on the wire or it could indicate there’s something else that’s not right. I’m guessing you’re not seeing call problems since it’s just on the RTCP stream and not the RTP stream.
Is there anything you did that correlates with the problem starting to happen (config change, asterisk update, phone firmware update, phone config change, etc)?
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Outside users calling outside the country? Fraud?
Take a look in extensions_custom.conf sip_custom.conf manager…etc if you’ve got anything weird or unexpected.
Why not if there’s some devices forwarded somewhere.
Check if your system accepts any anonymous sip and calls. It should not!
Think to update your system (O.S + FreePBX modules) as well!
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Issue remote extensión
Im working freepbx15 with asterisk 16.
Everything was going well, but suddenly that extension begins to have problems for incoming calls. (pjsip extension)
the system works like this
HOME 1:
-freepbx server
-wan FTTH with router without firewall
Voip open ports.
RTP open ports
LAN 192.168.1.0/24
HOUSE 2
WAN FTTH
LAN 192.168.0.0/24
Panasonic IP phone with configured extension wan-house1 voip port.
The extension connects correctly and can call, however, it cannot receive calls.
I have noticed in “report-asterisk info” that this remote extension registers with the ip of the internal lan.
258@192.168.0.20: 5060
when the right thing should be
258@WAN-IP.5060
What may be happening, has always connected well and I have not modified anything.
I appreciate your help in advance.
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Sngrep with tls support
TLS support in sngrep is limited: https://github.com/irontec/sngrep/issues/59
I realize that’s an old ticket but newer tickets about TLS decryption “not working” are referring back to that one.
As I understand it, this won’t let you capture TLS 1.2. I tested it out in my home lab. Configuring FreePBX to allow TLS 1.0 and up, only my ancient Obi 110 set for TLS shows up in the capture.
If they’re using modern forms of TLS in order to meet requirements, sngrep will probably not be useful in decryption mode. Other ideas might be to use Asterisk’s pjsip logger / sip debug, or have Asterisk do EEP mirroring (https://github.com/sipcapture/homer/wiki/Examples%3A-Asterisk) which you can then pick up with sngrep.
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Sngrep with tls support
I’m hoping that perhaps within the next year we’ll have the ability to just output pcap files via an option in the PJSIP packet logger, before decryption and encryption occurs, to make it easier to look at things. I don’t know if it’ll happen though but I’d like to see it.
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GraphQL: is it possible to "mutate" an extension's user name
Hello,
I would appreciate the ability to change one extension’s user name using FreePBX GraphQL or REST API.
Is it possible to do this ?
Doc [1] does not mention this possibility.
[1] https://wiki.freepbx.org/display/FPG/API
Best regards
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Outside users calling outside the country? Fraud?
Another important thing to check is your voicemail PINs. Fraudsters will sometimes leave a voicemail from a spoofed international number, then use weak voicemail PINs to call back to the number that left the voicemail through the voicemail system.
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Res_srtp.c: SRTCP unprotect failed because of authentication failure
If using S phones ensure they are on the latest firmware available as a fix for SRTP calls is included in it.
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Outside users calling outside the country? Fraud?
danardf:
extensions_custom.conf is blank. Same with sip_custom.conf. I do not allow anonymous sip calls. I do have some module updates to do but I was hoping to test the issue by doing what these people were doing and then patching to make sure that fixed it. I still can’t figure out from the logs what they are doing.
jsmith:
I don’t think I provide voicemail access from the outside. And since I don’t expose SIP to the outside, I dont’ think that could be happening.
Did they dial something special to get inside and then be able to dial any number? I’m having a hard time following the CDR report. Would me posting the asterisk logs be more helpful?
Below is the log from the day they were making the calls. And here are the bad numbers they dialed.
8769263700
8764100943
8766191578
4734028440
6493477814
2687700939
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All phones show as offline; outbound calls work, but inbound calls go straight to voicemail
Over the weekend, I turned off the server and the phones to perform a rack swap. When the phones and server were turned back on, outbound calls can be made, but inbound calls go straight to voicemail. The server is a FreePBX 60 system and the phones are Sangoma 505 and 3 of those have extenders.
The graph on the dashboard shows that all the users are offline, but show as online when they make an outbound call:
After seeing that, I checked the Chan_PJSip Endpoints section under Reports -> Asterisk Info -> chan_pjsip Info and confirmed that all the extensions show as unavailable. So since none of the phones in the ring group are available, the external calls go straight to voicemail. Trying to call from one phone to another also goes to that extension’s voicemail.
I’ve tried restarting the phones and the server, nothing changes. There are no new firewall rules since this started and I can ping the phones while logged in via ssh. Checking the asterisk log, the only thing that jumps out at me is this:
[2020-01-13 15:14:57] NOTICE[18819] res_pjsip_exten_state.c: Endpoint ‘117’ state subscription failed: Extension ‘*97’ does not exist in context ‘from-internal’ or has no associated hint
[2020-01-13 15:14:57] NOTICE[18819] res_pjsip_exten_state.c: Endpoint ‘117’ state subscription failed: Extension ‘*88’ does not exist in context ‘from-internal’ or has no associated hint
I get a similar message for each extension, but, I’m not sure if it is relevant.
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All phones show as offline; outbound calls work, but inbound calls go straight to voicemail
Those log messages may not be relevant, according to this: Sangoma Phone - PJSIP Registration Issue
Subscriptions are for BLFs and have nothing to do with registrations.
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