Welp. I went to the extreme. I wiped the edge router config and BAM - they register right off the bat.
Luckily, there weren’t many deltas between the backup and factory so we’ll see what starts fighting me. They also told me that the VPN service stopped at the same time so I’m wondering if they don’t just need a new router all together.
Thank you, @lgaetz.
Biggest pita I recall was they shipped in route mode and lan side did not have dhcp enabled or something like that
There there are dtmf sequences that can take it out of route mode as well and to enable dhcp - otherwise you have to go through a few gyrations to admin via browser
Once out of route mode? they behaved well with option 66 and picked up epm configs
If you want to use a profile rule manually the format I have noted is:
http://my.pbx.iporhost:84/spa$MA.xml
(Assuming http prov is set up on 84)
Admin guide for internal IVR config …
(Options 201 and 101 are the items I recall)
Hello, everyone.
I have read an article about Fast AGI and I would like to test it. What must I install on a remote machine (I have Gentoo Linux) in order that asterisk call remote functions? And another question is how I should process function results. In plain AGI they are returned via STDOUT.
You don’t need to install anything apart from a password-less tcp access to the caller. Of course you will need a process to service the connection, the result would be returned on stdout for asterisk to parse.
What process show I use? Can it be a call to an Apache web server script or it has to be a different software?
It can be anything. You are in control. Ask a question, wait for the answer. . . ( rm -rf / would not be a good thing if allowed 
)
I manually convert recording file from wav to opus and replace name in cdr table. But when i try listening this recording in CDR Reports i receive error message:
Unable to find an intermediary converter for /var/spool/asterisk/monitor/2019/10/03/out-1111-2222-20191003-133459-1570084499.25314.opus
File:/var/www/html/admin/libraries/media/Media/Media.php:299
Version FreePBX 14.0.5.2
@reraikes Thanks so much for the reply!
My issue has nothing to do with the IP changing, it has to do with the fact that the correct wan/Internet/Public IP is shown at some point in time after the change in the DDNS setting under Sys Admi. The issue is that the FreePBX DDNS client is not updating in any form of a timely manner. After days of waiting, I did a force be manually entering the same and correct, IP address and submitting it. Then in an hour or two the DNS servers propagated the new IP.
It just has to do with the system not updating the Sangoma server or whatever server is suppossed to update the DNS out in the digital world.
Sometimes I have a hard time describing my issue and hopefully this helps.
Oh… we are also on the most current version SysAdmin Pro and FreePBX distro.
John
That would be because there’s no support for it to be used in the GUI for playback.
I manage a few dozen system and am aware of the usual suspects that cause poor voice quality. Typical scenario has SIP endpoints at one place, and PBX elsewhere in a major datacenter. Real problems are rare, and some people exaggerate more than others, so it’s hard to tell if there really is a problem or not in some cases with certain people. Obviously it will never be perfect if you’re traversing public internet, and I do set that expectation.
I’m looking for a way to track/report on call quality. Is there such a way/tool at the asterisk level or within fpbx? Ideally a simple report/graph like the other server resource utilization on the main dashboard screen within fpbx, so I can get an overview of what a day looked like, vs sitting there staring at a trace for potentially hours on end.
FPBX distro 14.0.13.4 and Asterisk 13.22 with pjsip and usually g.711 end to end.
Which version of FreePBX?
Does anyone know about this please?
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thank you @Stewart1 for your reply, I have edited the outbound proxy as you suggest, but I didn’t understand where to place in my configuration the srvmi.p.ims.tiscali.net . Anyway now the error is different, I’m receiving the following
[2019-10-04 08:06:19] WARNING[1807]: res_pjsip_outbound_registration.c:993 handle_registration_response: 403 Forbidden fatal response received from 'sip:ims.tiscali.net:5060' on registration attempt to 'sip:0039XXXXXXXXXX@ims.tiscali.net:5060', retrying in '10' seconds
<--- Transmitting SIP request (623 bytes) to UDP:213.205.21.8:5060 --->
REGISTER sip:core1.p.ims.tiscali.net:5060 SIP/2.0
Via: SIP/2.0/UDP 82.84.72.XX:5060;rport;branch=z9hG4bKPje8eeb917-0188-4f0b-ac17-e85c305a3f13
From: <sip:0039XXXXXXXXXX@ims.tiscali.net>;tag=d48ea4b8-88b1-49fd-8944-c5e79b57d735
To: <sip:0039XXXXXXXXXX@ims.tiscali.net>
Call-ID: e4462ce5-b0b2-4d17-8df0-fcbbe2506ae9
CSeq: 20364 REGISTER
Contact: <sip:s@82.84.72.33:5060;line=qzyzcgz>
Expires: 3600
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Route: <sip:ims.tiscali.net:5060>
Max-Forwards: 70
User-Agent: FPBX-14.0.13.4(13.26.0)
Content-Length: 0
<--- Transmitting SIP request (503 bytes) to UDP:213.205.21.8:5060 --->
OPTIONS sip:core1.p.ims.tiscali.net:5060 SIP/2.0
Via: SIP/2.0/UDP 82.84.72.XX:5060;rport;branch=z9hG4bKPjcc61d4dc-1c5f-4dcc-85b9-61bb5a85ef70
From: <sip:0039XXXXXXXXXX@192.168.1.99>;tag=a30fc92e-c57b-412f-89a1-4d33c42dfc86
To: <sip:0039XXXXXXXXXX@ims.tiscali.net>
Contact: <sip:0039XXXXXXXXXX@82.84.72.33:5060>
Call-ID: 67701b88-74d2-40b9-9e53-13af39265a99
CSeq: 20100 OPTIONS
Route: <sip:0039XXXXXXXXXX@ims.tiscali.net:5060>
Max-Forwards: 70
User-Agent: FPBX-14.0.13.4(13.26.0)
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your log only shows you sending a REGISTER and OPTIONS to the core1.p.ims.tiscali.net which resolved to 213.205.21.8
doesnt look like they are responding
that first line refers to something which is not in your log
exactly, there is no response in the log… I cannot figure out this
there’s lines prior to all this, related to that warning with 403 forbidden, what are they ?
if your proxy is not responding to REGISTER, you’re using the wrong proxy or they are ignoring you