Was this a FreePBX based document or an Asterisk based document? There will be a big difference. Some providers have been know to provide Asterisk dialplan for interoping. That should not be confused with the already generated FreePBX dialplan which may make some of their documentation moot.
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Putting in a big dialplan etc
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Multi SIP trunk problem with inbound calls from same FreePbx
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FREE PBX backup and restore on another PBX
are these backups and restores stables? The Asterisk dies after 5 minutes after restore from another PBX. I am doing 4th time installation and always after restore it dies. Different hardware but for PBX restore that does not matter. Is there any known issue? Both are updated to the latest one.
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Multiple custom extensions for sending emails when called
Damn, your right!! Never thought about that (looking for the correct emoticon here . . .)
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Multiple custom extensions for sending emails when called
Here are some recommendations: 
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The Beta Release of FreePBX 15
@tm1000 Andrew am I missing something here? I do not see PBX Upgrader in Module Admin.
Current Asterisk Version: 16.3.0
FreePBX 14.0.8.4
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New build, same as the other builds, crashing every few hours
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Flowroute - Host based authentication
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PJSIP AOR goes offline every 2 minutes
I have one extension (it’s a mobile phone) that goes offline / online every 2 minutes. I have lots of other extensions that do not have this issue, including mobile phones that are the same model.
Anyway, relevant settings for her exten are:
x226
qualify: 60
max expiration: 7200
min expiration: 60
This is what the log looks like:
[2019-04-12 18:10:22] VERBOSE[31106] res_pjsip/pjsip_options.c: Contact 219/sip:219@10.0.10.219:5060 is now Reachable. RTT: 227.733 msec
[2019-04-12 18:11:19] VERBOSE[2093] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Reachable
[2019-04-12 18:11:19] VERBOSE[2093] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Reachable. RTT: 1410.088 msec
[2019-04-12 18:34:20] VERBOSE[2093] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Unreachable
[2019-04-12 18:34:20] VERBOSE[2093] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Unreachable. RTT: 0.000 msec
[2019-04-12 18:38:18] VERBOSE[23260] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Reachable
[2019-04-12 18:38:18] VERBOSE[23260] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Reachable. RTT: 573.735 msec
[2019-04-12 18:39:20] VERBOSE[14125] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Unreachable
[2019-04-12 18:39:20] VERBOSE[14125] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Unreachable. RTT: 0.000 msec
[2019-04-12 18:40:21] VERBOSE[10434] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Reachable
[2019-04-12 18:40:21] VERBOSE[10434] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Reachable. RTT: 619.236 msec
[2019-04-12 18:41:24] VERBOSE[14125] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Unreachable
[2019-04-12 18:41:24] VERBOSE[14125] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Unreachable. RTT: 0.000 msec
[2019-04-12 18:42:18] VERBOSE[4806] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Reachable
[2019-04-12 18:42:18] VERBOSE[4806] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Reachable. RTT: 985.925 msec
[2019-04-12 18:44:20] VERBOSE[30914] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Unreachable
[2019-04-12 18:44:20] VERBOSE[30914] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Unreachable. RTT: 0.000 msec
[2019-04-12 18:45:18] VERBOSE[23260] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Reachable
[2019-04-12 18:45:18] VERBOSE[23260] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Reachable. RTT: 1037.153 msec
[2019-04-12 18:55:20] VERBOSE[9411] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Unreachable
[2019-04-12 18:55:20] VERBOSE[9411] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Unreachable. RTT: 0.000 msec
[2019-04-12 18:57:17] VERBOSE[2093] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Reachable
[2019-04-12 18:57:17] VERBOSE[2093] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Reachable. RTT: 241.835 msec
[2019-04-12 18:58:20] VERBOSE[9411] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Unreachable
[2019-04-12 18:58:20] VERBOSE[9411] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Unreachable. RTT: 0.000 msec
[2019-04-12 18:59:19] VERBOSE[2093] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Reachable
[2019-04-12 18:59:19] VERBOSE[2093] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Reachable. RTT: 1471.551 msec
[2019-04-12 19:00:24] VERBOSE[14125] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Unreachable
[2019-04-12 19:00:24] VERBOSE[14125] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Unreachable. RTT: 0.000 msec
[2019-04-12 19:01:17] VERBOSE[31106] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Reachable
[2019-04-12 19:01:17] VERBOSE[31106] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Reachable. RTT: 55.421 msec
[2019-04-12 19:03:20] VERBOSE[10434] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Unreachable
[2019-04-12 19:03:20] VERBOSE[10434] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Unreachable. RTT: 0.000 msec
[2019-04-12 19:04:18] VERBOSE[31106] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Reachable
[2019-04-12 19:04:18] VERBOSE[31106] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Reachable. RTT: 423.966 msec
[2019-04-12 19:27:20] VERBOSE[2093] res_pjsip/pjsip_configuration.c: Endpoint 226 is now Unreachable
[2019-04-12 19:27:20] VERBOSE[2093] res_pjsip/pjsip_options.c: Contact 226/sip:226@[ip withheld]:6122;transport=TLS is now Unreachable. RTT: 0.000 msec
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Contact shows on Caller ID of screen
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Pause member in all queues after hanging up for 30 seconds (Not Wrapup time)
Is there a way using only the dial plan to automatically pause a member in all of their assigned queues for a fixed amount of time after hanging up a phone call and then unpause them? This is very similar to wrap up time, however wrap up time does not actually pause them and with the application I am building I need to be able to see the member paused when QueueMemberStatus is executed.
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Modules vulnerable to security threats have been automatically updated - framework has been automatically upgraded to fix security issues: SEC-2018-001
hey guys, any ideas on what to do if that fwconsole command didn’t solve the problem?
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Pause member in all queues after hanging up for 30 seconds (Not Wrapup time)
I figured it out. To anyone trying to accomplish the same thing this is the code I am using to do it.
$ext->add('macro-dialout-one-predial-hook', 's', '1', new \ext_noop('Entering user defined context macro-dialout-one-predial-hook in extensions_custom.conf'));
$ext->add('macro-dialout-one-predial-hook', 's', '', new \ext_setvar('CHANNEL(hangup_handler_push)','testing-hangup,s,1'));
$ext->add('testing-hangup', 's', '', new \ext_setvar('MemberChannel','Local/${CHANNEL(peername)}@from-queue/n'));
$ext->add('testing-hangup', 's', '', new \extension('PauseQueueMember(,${MemberChannel})'));
$ext->add('testing-hangup', 's', '', new \ext_system('sleep 5'));
$ext->add('testing-hangup', 's', '', new \extension('UnPauseQueueMember(,${MemberChannel})'));
$ext->add('testing-hangup', 's', '', new \ext_return());↧
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How does PBXact work with Zoho Unable to generate MOTD?
Hello：
I follow the link, but the system crashed. when I use USB to reinstall it, the sysem shows this error:
## ** CRITICAL SYSTEM ERROR ** Unable to generate MOTD. The /usr/sbin/fwconsole file is not accessible. You are likely to experience significant system issues
the fwconsole is in the directory, and I tried re install by USB many times, and every time it took few hours and I could not see any final indication to tell me the system have been installed successfully by USB. So, I have to plug up the USB after few hours and reboot the system. when I reboot it, the system always shows the error as above.
Do you have any suggestion?!
Reboot shows erros-Unable to generate MOTD:
During install step, always cursor and has no final indication:
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t38 Fax detection
Well, really sorry to warm up this old, old, thread. 
Here follows my conclusion how I have fixed that, and several other T.38 related issues, - maybe this will help also others with similar problems.
Regarding the T.38 fax detection, - the best solution is simply to not use it.
If you have a phone number dedicated to faxing-only then disable fax detection on the inbound route and send the call directly to the “Fax Recipient”. For me this had worked perfectly. By the way, it is also strongly recommended to disable any jitter buffer.
Another really problematic point is the correct setting of the T.38 parameters in FreePBX. For example, regarding T38 Pass-Through it is only possible to enable or disable it. Both of them were wrong in my situation. If I turned it on, the parameter t38pt_udptl=yes was established in the config files. Well, my IP phone & fax provider needs t38pt_udptl=yes,redundancy. So I left it disabled and set the t38pt_udptl=yes,redundancy function by hand below in “Other SIP Settings”. I also added here faxdetect=t38 and t38pt_usertpsource=yes. However, I am not sure about the latter one if that is really necessary.
To make the confusion perfect, these parameters also exist (with different syntax) incidentally in the udptl.conf file. Do not set it there, - these are deprecated! So far I understand the _udptl.con_f is only used regarding any udptl settings.
Furthermore, in legacy FreePBX 11, there seems to exist (sometimes) a grave installation bug regarding the Digium fax driver. No matter how many times “Free Fax For Asterisk” was “installed” through the FreePBX GUI (in Digium Addons), it was never installed in the modules directory. The res_fax_digium.so module was simply missing there. 
So I had to install it by hand from the digium website http://downloads.digium.com/pub/telephony/fax/res_fax_digium/
After this, to avoid any conflict, I have disabled the Spandsp FAX Driver in modules.conf with noload => res_fax_spandsp.so. It should be noted here that any newer FreePBX above 12 need this driver for faxing. Most have version 0.0.6 installed, - in my case this driver never worked. If someone has similar issues he should try any more recent version: https://www.soft-switch.org/downloads/spandsp/snapshots/
Finally, my current working fax trunk (FOIP) config looks so:
type=peer
secret=xxx
qualify=yes
insecure=port,invite
host=business2.voipgateway.org
fromdomain=business2.voipgateway.org
disallow=all
directmedia=yes
defaultuser=xxx
context=from-pstn-toheader
allow=alaw&ulaw
dtmfmode=rfc2833The only unclear parameter is directmedia=yes which is in most examples disabled, especially if NAT is used. Well, in my case (with NAT but opened UDPTL ports) it works with yes, so I let it enabled.
That’s it, - I can say without dramatizing, it took me several years to figure all this out. 
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Debug src ip ", "Received incoming SIP connection from unknown peer to 123") in new stack"
i dont understand .
why some calls come with src ip and other calls without src ip ?!!
i think this option comes from freepbx not asterisk .
do you think we can tune freepbx to display that src ip in logs ?
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Customer Support Line with Time Restrictions
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T38 Faxing and T38pt_udptl= values and devices #2
Well, - we have now 2019 and I am happy to note here that my FoIP (faxing over IP) is working meanwhile stable and reliable. 
It looks that my provider Sipcall.ch has resolved their T.38 issues. Furthermore I was able to find out the correct T.38 faxing parameters. And finally I have also discovered a strange “Free Fax For Asterisk” installation bug. Also this problem needed his time to be solved…
So after five years, this story has come finally to an end!
More detailed information can be found at the following thread:
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Cant access FreePBX with WinSCP
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Sangoma S500 Voicemail Button - Password?
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