Those are messages that are not related to your issue. It is just warning you that FreePBX is using deprecated functions and everybody using FreePBX distro is getting those same messages.
If you modified your network settings and somehow they ended misconfigured, you are probably going to experience this kind of issues.
The only thing that remains is on the dashboard, the Asterisk is stuck at 8 minutes 45 seconds runtime and is not updating. The system has been running for hours now
So what you mean is that the dashboard widgets are not refreshing? Unfortunately, that is a frequent issue and so far I haven’t found a solution for that. In my case, the Asterisk graphics only work for hour and day, but week and month are not working.
Yes, the dashboard widget for Asterisk is not updating at all. Stuck with the triangle about running less than 20 minutes and SysInfo updated -20769 seconds ago.
Looks like something went wonky as it’s showing a negative time for sysinfo
Not a super big deal, just gets my OCD a bit of a flare. The negative sysinfo is counting down so maybe it will start working when it gets a positive count
You can try rebooting the server, but yes, you will have to control your OCD for the time being 
Hello
I am confused on the difference between DID and CID. I know that DID stands for Direct in Dial and CID stands for Caller Id. I did google it on line but the examples that they put is kind of confusing… Can anyone please provide one good example??
Thank You
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Actually, if you read this poor guys post, “they won’t” and “they haven’t”, he has (unsuccessfully) tried to go bare-back but with 40+ options and a round trip of each one about two minutes, then he has yet a failed.
CID is not ever “generally” transmitted (but ALWAYS strictly by the the companies posted methodology, the trouble for this poor blighter is they refuse to divulge their methodology.) but back to the story, most all FSK or DTMF is apparent in the “dead space” between between the first and second “ring cadence” Australia and UK as ever are a little contrary.
This thread laid out a method that is very diagnostic, All DAHDI calls are “In-Band”, you can record and later analyse the audio, which gives you your first clue, “what did you hear?” an adjunct to that that can be confusing is that the analog line might and might not be “flashed” or even “reversed” before or after the “spill”, on a $5 dollar walmart phone you will see the dial pad light blink
JM2CWAE
Hello @all,
i did not found a solution to this, my problem, therefore i am asking here.
I cant do none assisted calls, the system will loose functionality to calls from outside.
*2 < ext > # does not work. (sorry brackets are gone, try it with space, textform: star two bracket ext bracket hash, with no space)
What is going wrong?
Example, what goes wrong. 
Outside call comes to a process/queue and so on, all that stuff works as designed. Also pressing 1/2/5 in IVR works fine. If nothing happens -nobody will answer the call, after X min. the system plays an announcement, and hangs up. That all is very fine.
The call flow ist fine.
If the queue members ring (211 or 212) -somebody get che call, we can do ## < ext > # many times. As playing football. The call will not get lost.
But no *2 < ext > # will work.
But …
After the first ext hangs up, the callee will not be connected to the new ext, instead, the callee will hear the music/MOH until he dies. 
But after this procedure, a new call will not ring on the queue members, on ext 211/212. They are dead for the outside call.
This is possible:
216 can call 211/212
211 -> 212 OK
212 -> 212 OK
But, if somebody calls from outside, none of the queue-members 211/212 will ring. There is NO timeout, that it will works after this timeout. Nope. Only a
/var/lib/asterisk/bin/amportal restart
will reset the system.
asterisk -rx “core show channels” shows:
Channel Location State Application(Data)
PJSIP/211-00000011 s@macro-dial-one:1 Up AppDial((Outgoing Line))
PJSIP/212-00000013 s@macro-dial-one:1 Up AppDial((Outgoing Line))
2 active channels
0 active calls
43 calls processed
Asterisk ending (0).
Assisted call transfer kills (somehow) the whole functionality (for the outside callee)
Other calls to ext 214/215 etc will ring. Also
Call to 214 (direct form outside) rings, ##<211># is OK.
Call to 214 (direct form outside) rings, *2# is also OK, as pressed *2 MOH will plays, the other rings, get the call, speak to 214, as above after 214 hangs up, MOH will play to callee, the call is not asssited transferd to other .
At this status, no intercom possible. Only
amportal restart
will clean/reset the system, brings (for a short period) the whole functionality, untel the first assisted transfers (which also will not works).
asterisk -rx “pjsip show endpoints” | grep -e ‘in|Unava’
Endpoint: 11/11 Unavailable 0 of inf - OK
Endpoint: 12/12 Unavailable 0 of inf - OK
Endpoint: 13/13 Unavailable 0 of inf - OK
Endpoint: 211/211 In use 2 of inf - why? QMember
Endpoint: 212/212 In use 1 of inf - why? QMember
Endpoint: 213/213 In use 1 of inf - why?
Endpoint: 214/214 On Hold 3 of inf - ?
Endpoint: 215/215 Not in use 0 of inf OK
Endpoint: 216/216 In use 2 of inf - why?
Endpoint: 217/217 Unavailable 0 of inf - OK/NC
Endpoint: 218/218 Not in use 0 of inf OK
Endpoint: 219/219 Not in use 0 of inf OK
Endpoint: 220/220 Not in use 0 of inf OK
Endpoint: 221/221 Not in use 0 of inf OK
Endpoint: 99/99 Not in use 0 of inf OK
After a time period:
asterisk -rx “core show channels”
Channel Location State Application(Data)
PJSIP/216-0000004d (None) Up AppDial((Outgoing Line))
PJSIP/216-00000053 (None) Up AppDial((Outgoing Line))
PJSIP/211-00000011 s@macro-dial-one:1 Up AppDial((Outgoing Line))
PJSIP/214-0000004c s@macro-dial-one:54 Up Dial(PJSIP/216/sip:216@IP-Addr
PJSIP/214-00000052 s@macro-dial-one:54 Up Dial(PJSIP/216/sip:216@IP-Addr
PJSIP/214-00000039 (None) Up AppDial((Outgoing Line))
Local/211@from-inter 211@from-internal-xf Up (None)
Local/211@from-inter s@macro-dial-one:54 Up Dial(PJSIP/211/sip:211@IP-Addr
PJSIP/211-0000003a (None) Up AppDial((Outgoing Line))
Local/213@from-inter s@macro-dial-one:54 Up Dial(PJSIP/213/sip:213@IP-Addr
Local/213@from-inter 213@from-internal-xf Up (None)
PJSIP/213-00000054 (None) Up AppDial((Outgoing Line))
PJSIP/212-00000013 s@macro-dial-one:1 Up AppDial((Outgoing Line))
13 active channels
4 active calls
132 calls processed
The HW is, YL 48S (211/217) /46S (212) /60p (213/4/5/6/8) with five 56H.
Tested also temp. with Grandstream 2170 as 211, same.
Any, any help appreciated. Thanks in advance.
ELindemann
From my understanding of this post and previous replies, after it rings an internal extension it is forwarding to an external number (your cellphone) you cannot control the ring time of your cellphone that is up to the carrier. You could however create a new extension and use Zoiper on your phone, you could control that ringtime.
@lgaetz I finally got it working. I’m not sure what I was doing wrong earlier but I can confirm it does work on FreePBX 14 with asterisk 13.22.
Those warning messages still show up but everything works the way it’s suppose to. So they can be ignored.
I missed that part, but it sounds too familiar. Back here in Argentina it is also VERY difficult to get any technical details from anyone, let alone 1st tier customer support representative.
Almost all technical info regarding telephony standards, especially back in the day when PRI service was famous here, before SIP trunking became famous, was coming from other technically savvy users who happened to have deep contacts inside the telephone company technical staff.
Here in Argentina, the standard is very mixed up. For example on POTS lines from the original national telephone company, Caller ID is ETSI FSK as in Europe, but impedance is 600 omhs as in USA. Then when telephone lines got deregulated, everybody started doing whatever they wanted without any user facing documentation whatsoever. VoIP providers that give you an FXS ATA set it to USA region, so you get common US parameters on the FXS port, like Bellcore Caller ID, US CP tones, and sometimes they also enable polarity reversal to signal call disconnection. So it is basically a lottery.
Don’t get me started on PRI, which varies according to the provider, the most “common” one being such a specific signalling configuration : E1 CAS MFCR2 with the specific MFCR2 Argentina variant.
You call your mother from your cell phone or your PBX extension, she sees your “Caller*ID” as the number of your cellphone or your PBX extension.
Your PBX gets a call from your mother, the DID is the number you told here to call you and the Caller*ID is the number of her phone.
I feel your pain, given your situation, under no circumstances is the FreePBX DAHDI “helper” ever going to “help you” it knows little about anything but Generic US PRI’s never mind fifty years of RBS (T1’s) and even has trouble with euroisdn , move on . . . DAHDI itself outside of FreePBX is more than capable to handle SS7, MFCR2, Ameteur radio, alarmco, dynamic-eth and all sorts of even more esoteric channels
I currently use Bria Teams for softphones which does not have to create the "90"EXT extensions and we use PJSIP. I’ve been meaning to give Zulu a run because Bria is expensive at $4-$5 per month a user. So currently about 3K per year in licence fees Ouch!.
I haven’t had any issues getting pjsip to work with FOP2 or auto creating buttons. However have you tried creating a custom button for the 90EXT? Even if you look at asterisk info peers you’ll likely see a separate registration for 90EXT. So that maybe what FOP2 needs to target.
You might also check in Freepbx advanced settings whether WebRTC uses sip,pjsip or auto. FOP2 recently also just came out with it’s own built in softphone which I never played with which uses webrtc. I’m sure zulu is much better.
Thank you so much. Your example really did help i really appreciate it.
I have no issues with this. Follow me rings for the exact amount of time. I should note though that a call out to a cell phone could take 5 seconds or more before it actually starts ringing the cell phone due to post dial delay on the outbound carrier. 8 seconds is way too short, and don’t use 8 seconds lol. Each ring is 5 seconds. So 8 seconds is a ring and a half and it’s weird. Try 10 to extension and 15 to cell and see if that helps anything.
What is your ring strategy and where is your follow me no answer destination? Is it to a voicemail? Or is it set to normal routing? Try setting that to go to a VM.
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Thanks for your help so far on this!
I had it working for a couple of days and has now stopped working. When trying to dial from the Mitel to FreePBX I get a busy signal and the following error message:
[2019-08-29 21:17:50] WARNING[9401][C-00000013]: chan_sip.c:17266 check_auth: username mismatch, have <5001>, digest has <>
[2019-08-29 21:17:50] NOTICE[9401][C-00000013]: chan_sip.c:26364 handle_request_invite: Failed to authenticate device "BEDSIDE" <sip:8286930479@192.168.0.3:5160;transport=UDP>;tag=52b8a68-7f000001-13c4-e6a7d-cef3a76d-e6a7d
5001 is one of the FXS ports that is on the Adtran that I also have registered to the FreePBX. The FXS ports are working correctly. However, the PRI is now giving me the above message. The only way that I have been able to fix this is to delete both the trunk and outbound route from FreePBX and then recreate them. Once I do that it seems to work for a day or so before it starts throwing this error again.
Below is the config of the trunk in FreePBX:
General Tab:
Trunk Name: Mitel
Hide Caller ID: No
Outbound Caller ID: “out” (main DID)
CID Options: Allow Any CID
Max Channels: (blank)
Asterisk Trunk Dial Options: T (system)
Continue if busy: No
Disable Trunk: No
Monitor Trunk Failures: No
Dialed Number Manipulation Rules:
I have a couple of rules to convert DIDs to 4 digits from 10
SIP Settings Tab:
Outgoing Tab:
Trunk Name: PRI_to_Mitel
Peer Details:
disallow=all
allow=ulaw
canreinvite=no
dtmfmode=rfc2833
host=192.168.0.9
insecure=port,invite
port=5060
qualify=yes
type=friend
context=from-internal
match_auth_username=no
Incoming tab is all blank.
Hoping someone will be able to help with this. I have the trunks in FreePBX setup as CHAN_SIP and the extensions that the FXS ports are registering to are also using CHAN_SIP. If it would be better to convert everything over to PJSIP I am happy to do so, but I may need a little guidance on what settings to use as I was not able to figure it out when I tried the other day.
Thanks,
Daniel
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