@thepossum905 wrote:

I have a fairly new freepbx 14 installation, the phones are on a remote network from the hosted pbx. Generally things are running swimmingly, although we’ve recently just enabled g729 in order to save on bandwidth.

There are some intermittent network issues with the internet provider, causing occasional call quality issues. All the boring stuff yup, a bit of extra packet loss and jitter that we can’t predict or prevent.
So we need to somehow visualize the per-call quality, in some sort of automated manner.

From the asterisk CLI, I can manually “pjsip show channelstats” and see live info on current active calls. Or I can script a “asterisk -rx ‘pjsip show channelstats’” to see everything at a periodic interval.

But what would be much more helpful would be for asterisk to log the per-call stats at the end of each call.

My google fu repeatedly brings up a command “rtcp set stats on”, but … nothing changes in the logs. How can I see the produced jitter data so that I can start to visualize it?

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@asterisk011 wrote:

Hey, I’ve just installed FreePBX and Asterisk on my Centos 7 VM.
When I entered the FreePBX GUI I got an error saying:

Reload failed because retrieve_conf encountered an error: 1

exit: 1
Unable to continue. Error with the templating engine: array_merge(): Argument #2 is not an array:: in /var/www/html/admin/libraries/utility.functions.php on line 207 #0 /var/www/html/admin/libraries/BMO/Less.class.php(209): die_freepbx(‘Error with the …’) #1 /var/www/html/admin/libraries/BMO/Less.class.php(94): FreePBX\Less->getCachedFile(’/var/www/html/a…’, ‘/admin/assets/f…’, Array) #2 /var/lib/asterisk/bin/retrieve_conf(172): FreePBX\Less->generateModuleStyles(‘featurecodeadmi…’) #3 /var/lib/asterisk/bin/retrieve_conf(550): connectdirs->generate_less(‘featurecodeadmi…’) #4 {main}

Plus, the Please Select the default locales of the PBX page keeps showing up, when I click on Submit I get Sign me up window and it’s an idle loop. Can’t configure anything. Please help.

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@jamesg224 wrote:

Background info:
I’ve been running FreePBX for just over a year now and it has been incredibly stable with SIP trunks (no POTS). There are no ports forwarded as I am using SIP registration. Established/related traffic is allowed for time/system/firmware updates etc. Otherwise everything is on a private LAN behind a firewall.

Question:
This week I have been asked to connect a POTS line so that anyone who calls this legacy number will still get through until the contract runs out. Previously we were just using one separate phone. During testing the system connectivity between the SPA-3102 and the PBX, everything is fine. As soon as I connect the POTS line to the SPA-3102, I am receiving IPS messages from my firewall (current Unifi USG). This has occurred twice, both times about 10 minutes after connecting the SPA to the line. Since I have started using FreePBX, there have been no other IPS hits so this is new behaviour.

This is my IPS message:
Message: IPS Alert 2: Misc Attack. Signature ET TOR Known Tor Relay/Router (Not Exit) Node Traffic group 276. From: 193.224.163.43:11371, to: mypbxaddress:48860, protocol: TCP

As far as I know port 48860 is not in use for anything so I’m puzzled as to what this could be (it could also just be a false positive!). Any input would be greatly appreciated. Thanks.

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@thepossum905 wrote:

I have a fairly new freepbx 14 installation, the phones are on a remote network from the hosted pbx. Generally things are running swimmingly, although we’ve recently just enabled g729 in order to save on bandwidth.

There are some intermittent network issues with the internet provider, causing occasional call quality issues. All the boring stuff yup, a bit of extra packet loss and jitter that we can’t predict or prevent.
So we need to somehow visualize the per-call quality, in some sort of automated manner.

From the asterisk CLI, I can manually “pjsip show channelstats” and see live info on current active calls. Or I can script a “asterisk -rx ‘pjsip show channelstats’” to see everything at a periodic interval.

But what would be much more helpful would be for asterisk to log the per-call stats at the end of each call.

My google fu repeatedly brings up a command “rtcp set stats on”, but … nothing changes in the logs. How can I see the produced jitter data so that I can start to visualize it?

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@bksales wrote:

Have a customer that wants very specific feature codes for flow controls that overlap with the existing ones. As in they basically want to reorder *280-*284. I looked in the database and see a column for defaultcode and customcode. What’s the easiest way to change this? When I try from the GUI it warns me that it already exists even if I change them to something else completely first (I made *380-*384).

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@thepossum905 wrote:

I have a fairly new freepbx 14 installation, the phones are on a remote network from the hosted pbx. Generally things are running swimmingly, although we’ve recently just enabled g729 in order to save on bandwidth.

There are some intermittent network issues with the internet provider, causing occasional call quality issues. All the boring stuff yup, a bit of extra packet loss and jitter that we can’t predict or prevent.
So we need to somehow visualize the per-call quality, in some sort of automated manner.

From the asterisk CLI, I can manually “pjsip show channelstats” and see live info on current active calls. Or I can script a “asterisk -rx ‘pjsip show channelstats’” to see everything at a periodic interval.

But what would be much more helpful would be for asterisk to log the per-call stats at the end of each call.

My google fu repeatedly brings up a command “rtcp set stats on”, but … nothing changes in the logs. How can I see the produced jitter data so that I can start to visualize it?

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@imgoemon wrote:

Why do we use port ranged 10000-20000 for RTP instead of port 5060 ?

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@thepossum905 wrote:

I have a fairly new freepbx 14 installation, the phones are on a remote network from the hosted pbx. Generally things are running swimmingly, although we’ve recently just enabled g729 in order to save on bandwidth.

There are some intermittent network issues with the internet provider, causing occasional call quality issues. All the boring stuff yup, a bit of extra packet loss and jitter that we can’t predict or prevent.
So we need to somehow visualize the per-call quality, in some sort of automated manner.

From the asterisk CLI, I can manually “pjsip show channelstats” and see live info on current active calls. Or I can script a “asterisk -rx ‘pjsip show channelstats’” to see everything at a periodic interval.

But what would be much more helpful would be for asterisk to log the per-call stats at the end of each call.

My google fu repeatedly brings up a command “rtcp set stats on”, but … nothing changes in the logs. How can I see the produced jitter data so that I can start to visualize it?

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@JerryFernandez wrote:

Hi,

I’m testing the Pinset module in FreePBX 14.0.13.4 and during my test, when using IP Phone it accept the pinset but when using the softphone(zoiper 2.8.39) even though I inputted the correct pinset it still saying “password incorrect”. Then when I check the log it show “user entered nothing”, I’m not sure why it show this error even if I inputted the pinset.

This is the log when I use softphone
– Executing [DESTINATION_NUMBER@from-internal:3] Macro(“SIP/5555-00000d48”, “pinsets,1,0”) in new stack
– Executing [s@macro-pinsets:1] Set(“SIP/5555-00000d48”, “try=1”) in new stack
– Executing [s@macro-pinsets:2] GotoIf(“SIP/5555-00000d48”, “0?cdr,1”) in new stack
– Executing [s@macro-pinsets:3] GotoIf(“SIP/5555-00000d48”, “1?auth:return”) in new stack
– Goto (macro-pinsets,s,4)
– Executing [s@macro-pinsets:4] Progress(“SIP/5555-00000d48”, “”) in new stack
– Executing [s@macro-pinsets:5] Read(“SIP/5555-00000d48”, “dtmf,agent-pass,0,n,1,10”) in new stack
– <SIP/5555-00000d48> Playing ‘agent-pass.ulaw’ (language ‘en’)
– User entered nothing.
– Executing [s@macro-pinsets:6] GotoIf(“SIP/5555-00000d48”, “0?return:askpin”) in new stack
– Goto (macro-pinsets,s,7)
– Executing [s@macro-pinsets:7] Set(“SIP/5555-00000d48”, “try=2”) in new stack
– Executing [s@macro-pinsets:8] GotoIf(“SIP/5555-00000d48”, “0?hangup”) in new stack
– Executing [s@macro-pinsets:9] Read(“SIP/5555-00000d48”, “dtmf,auth-incorrect,0,n,1,10”) in new stack
– <SIP/5555-00000d48> Playing ‘auth-incorrect.ulaw’ (language ‘en’)
– User entered nothing.

This is the log when I use IP Phone
– Executing [DESTINATION_NUMBER@from-internal:3] Macro(“SIP/7000-00000d3a”, “pinsets,3,0”) in new stack
– Executing [s@macro-pinsets:1] Set(“SIP/7000-00000d3a”, “try=1”) in new stack
– Executing [s@macro-pinsets:2] GotoIf(“SIP/7000-00000d3a”, “0?cdr,1”) in new stack
– Executing [s@macro-pinsets:3] GotoIf(“SIP/7000-00000d3a”, “1?auth:return”) in new stack
– Goto (macro-pinsets,s,4)
– Executing [s@macro-pinsets:4] Progress(“SIP/7000-00000d3a”, “”) in new stack
– Executing [s@macro-pinsets:5] Read(“SIP/7000-00000d3a”, “dtmf,agent-pass,0,n,1,10”) in new stack
– <SIP/7000-00000d3a> Playing ‘agent-pass.ulaw’ (language ‘en’)
– User entered ‘1234’
– Executing [s@macro-pinsets:6] GotoIf(“SIP/7000-00000d3a”, “1?return:askpin”) in new stack
– Goto (macro-pinsets,s,12)
– Executing [s@macro-pinsets:12] NoOp(“SIP/7000-00000d3a”, “returning back”) in new stack

Any Ideas/Help would be greatly appreciated

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@doubleview wrote:

Hi all,

I have a custom extension which if called needs to send an email to an address.

I’ve had this solution working for a while but it seems to have stopped working now…no emails are being sent, tried using a different address.

Definitely not a spam folder issues either.

[send-email]
exten => s,1,NoOp(Entering user defined context [send-email] in extensions_custom.conf)
exten => s,1,System(echo ‘Call from ${CALLERID(name)} at ${CALLERID(number)}’ | mail -s ‘SUBJECT’ ‘email@gmail.com’)
exten => s,n,hangup()

Dial field on the custom extension under the advanced tab is: local/s@send-email

Checked the firewall, everything seems ok, freepbx is sending admin alert emails, and it also works if you manually enter the commands into the command prompt

echo ‘Call from ${CALLERID(name)} at ${CALLERID(number)}’ | mail -s ‘SUBJECT’ ‘email@gmail.com’

Just wondering iff anyone could help me.

Thanks.

Asterisk Log

[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [state-not_set@sub-presencestate-display:1] Set(“SIP/2000-00000011”, “PRESENCESTATE_DISPLAY=”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [state-not_set@sub-presencestate-display:2] Return(“SIP/2000-00000011”, “”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-dial-one:47] Set(“SIP/2000-00000011”, “CONNECTEDLINE(name,i)=pager”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-dial-one:48] Set(“SIP/2000-00000011”, “CONNECTEDLINE(num)=9005”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-dial-one:49] Set(“SIP/2000-00000011”, “D_OPTIONS=TtrI”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-dial-one:50] Macro(“SIP/2000-00000011”, “dialout-one-predial-hook,”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-dialout-one-predial-hook:1] MacroExit(“SIP/2000-00000011”, “”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-dial-one:51] ExecIf(“SIP/2000-00000011”, “0?Set(D_OPTIONS=trII)”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-dial-one:52] NoOp(“SIP/2000-00000011”, “”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-dial-one:53] ExecIf(“SIP/2000-00000011”, “0?Set(D_OPTIONS=TtrIg)”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-dial-one:54] Dial(“SIP/2000-00000011”, “local/s@send-email,TtrIb(func-apply-sipheaders^s^1)”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] app_stack.c: Local/s@send-email-00000003;1 Internal Gosub(func-apply-sipheaders,s,1) start
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@func-apply-sipheaders:1] NoOp(“Local/s@send-email-00000003;1”, “Applying SIP Headers to channel Local/s@send-email-00000003;1”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@func-apply-sipheaders:2] Set(“Local/s@send-email-00000003;1”, “TECH=Local”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@func-apply-sipheaders:3] Set(“Local/s@send-email-00000003;1”, “SIPHEADERKEYS=”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@func-apply-sipheaders:4] While(“Local/s@send-email-00000003;1”, “0”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] app_while.c: Jumping to priority 12
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@func-apply-sipheaders:13] Return(“Local/s@send-email-00000003;1”, “”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] app_stack.c: Spawn extension (send-email, 9005, 1) exited non-zero on ‘Local/s@send-email-00000003;1’
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] app_stack.c: Local/s@send-email-00000003;1 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
[2019-09-30 07:32:32] VERBOSE[6895][C-0000000a] pbx.c: Executing [s@send-email:1] NoOp(“Local/s@send-email-00000003;2”, “Entering user defined context [send-email] in extensions_custom.conf”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] app_dial.c: Called local/s@send-email
[2019-09-30 07:32:32] VERBOSE[6895][C-0000000a] pbx.c: Executing [s@send-email:2] System(“Local/s@send-email-00000003;2”, “echo ‘Call from davids Desk at 2000’ | mail -s ‘SUBJECT’ ‘email@gmail.com’”) in new stack
[2019-09-30 07:32:32] VERBOSE[6895][C-0000000a] pbx.c: Executing [s@send-email:3] System(“Local/s@send-email-00000003;2”, “echo ‘Call from davids Desk at 2000’ | mail -s ‘SUBJECT’ ‘email@gmail.com’”) in new stack
[2019-09-30 07:32:32] VERBOSE[6895][C-0000000a] pbx.c: Executing [s@send-email:4] Hangup(“Local/s@send-email-00000003;2”, “”) in new stack
[2019-09-30 07:32:32] VERBOSE[6895][C-0000000a] pbx.c: Spawn extension (send-email, s, 4) exited non-zero on ‘Local/s@send-email-00000003;2’
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] app_dial.c: No one is available to answer at this time (1:0/0/0)
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-dial-one:55] ExecIf(“SIP/2000-00000011”, “0?MacroExit()”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-dial-one:56] ExecIf(“SIP/2000-00000011”, “0?Set(DIALSTATUS=)”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-dial-one:57] GosubIf(“SIP/2000-00000011”, “0?s-NOANSWER,1()”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-dial-one:58] MacroExit(“SIP/2000-00000011”, “”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-exten-vm:27] Set(“SIP/2000-00000011”, “SV_DIALSTATUS=NOANSWER”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-exten-vm:28] GosubIf(“SIP/2000-00000011”, “0?docfu,1()”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-exten-vm:29] GosubIf(“SIP/2000-00000011”, “0?docfb,1()”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-exten-vm:30] Set(“SIP/2000-00000011”, “DIALSTATUS=NOANSWER”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-exten-vm:31] ExecIf(“SIP/2000-00000011”, “0?MacroExit()”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-exten-vm:32] GotoIf(“SIP/2000-00000011”, “1?s-NOANSWER,1”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx_builtins.c: Goto (macro-exten-vm,s-NOANSWER,1)
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s-NOANSWER@macro-exten-vm:1] GotoIf(“SIP/2000-00000011”, “0?exit,1”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s-NOANSWER@macro-exten-vm:2] PlayTones(“SIP/2000-00000011”, “congestion”) in new stack
[2019-09-30 07:32:32] WARNING[6892][C-0000000a] translate.c: no samples for alawtolin
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s-NOANSWER@macro-exten-vm:3] Congestion(“SIP/2000-00000011”, “10”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] app_macro.c: Spawn extension (macro-exten-vm, s-NOANSWER, 3) exited non-zero on ‘SIP/2000-00000011’ in macro ‘exten-vm’
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Spawn extension (from-internal, 9005, 2) exited non-zero on ‘SIP/2000-00000011’
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [h@from-internal:1] Macro(“SIP/2000-00000011”, “hangupcall”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/2000-00000011”, “1?theend”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/2000-00000011”, “0?Set(CDR(recordingfile)=)”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“SIP/2000-00000011”, " montior file= /var/spool/asterisk/monitor/2019/09/30/internal-9005-2000-20190930-073231-1569828751.23.wav") in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“SIP/2000-00000011”, “1?skipagi”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-hangupcall:7] Hangup(“SIP/2000-00000011”, “”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘SIP/2000-00000011’ in macro ‘hangupcall’
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [h@from-internal:1] Macro(“SIP/2000-00000011”, “hangupcall”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/2000-00000011”, “1?theend”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/2000-00000011”, “0?Set(CDR(recordingfile)=)”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“SIP/2000-00000011”, " montior file= /var/spool/asterisk/monitor/2019/09/30/internal-9005-2000-20190930-073231-1569828751.23.wav") in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“SIP/2000-00000011”, “1?skipagi”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Executing [s@macro-hangupcall:7] Hangup(“SIP/2000-00000011”, “”) in new stack
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘SIP/2000-00000011’ in macro ‘hangupcall’
[2019-09-30 07:32:32] VERBOSE[6892][C-0000000a] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/2000-00000011’
[2019-09-30 07:32:32] VERBOSE[6894][C-0000000a] app_mixmonitor.c: MixMonitor close filestream (mixed)
[2019-09-30 07:32:32] VERBOSE[6894][C-0000000a] app_mixmonitor.c: End MixMonitor Recording SIP/2000-00000011

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@thepossum905 wrote:

I have a fairly new freepbx 14 installation, the phones are on a remote network from the hosted pbx. Generally things are running swimmingly, although we’ve recently just enabled g729 in order to save on bandwidth.

There are some intermittent network issues with the internet provider, causing occasional call quality issues. All the boring stuff yup, a bit of extra packet loss and jitter that we can’t predict or prevent.
So we need to somehow visualize the per-call quality, in some sort of automated manner.

From the asterisk CLI, I can manually “pjsip show channelstats” and see live info on current active calls. Or I can script a “asterisk -rx ‘pjsip show channelstats’” to see everything at a periodic interval.

But what would be much more helpful would be for asterisk to log the per-call stats at the end of each call.

My google fu repeatedly brings up a command “rtcp set stats on”, but … nothing changes in the logs. How can I see the produced jitter data so that I can start to visualize it?

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@FreerPBXer wrote:

I’m reposting this from a locked thread because it didn’t get an answer.

I’ve read both these articles, just to preface.

https://wiki.freepbx.org/display/FPG/Time+Groups+Sample+Configurations
https://wiki.freepbx.org/display/FPG/Time+Group+User+Guide

In order to ease holiday scheduling we switched to defining non-working hours. And the two articles don’t say what happens if the start time for a time condition is after the end time (i.e. start at 5:00 PM and end at 8:00 AM). And if the end day for such a condition is Thursday, does it end with the change from positive match to negative at 8:00 AM, or from negative to positive at 5:00 PM? Similarly, with Monday would it start with a neg match at 8:00 AM or with a positive at 17:00?

If we need to define separately as 00:00 to 08:00 and also 17:00 to 23:59, then it seems like there would be a one minute unmatched state every night from 23:59 to 00:00.

Lastly, what if there is a conflict? i.e. we could define 08:00 to 17:00 M-F as work hours and set call flow based on that. But if we define a Monday as a holiday and designate after-hours call flow, which group ‘wins’?

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@blauau wrote:

Hello!
I have a Nokia 2.1 DualSim phone, connected to RasPBX (Asterisk 13.27.0) via BT. The connection still works (was not sure about rfcomm port number so i leaved default 3), but the calls will not forwarded to my extension (1000, sip extension).
Mobile conf looks like:
[general]
interval=30

[adapter]
id=blue
address=00:02:72:D2:XX:XX

[Nokla21]
address=AC:57:75:2E:XX:XX
port=3
context=from-internal
adapter=blue
group=1

raspbx*CLI> mobile show devices
ID Address Group Adapter Connected State SMS
Nokla21 AC:57:75:2E:XX:XX 1 blue Yes Free No

By the inbound routes i use ANY for CID and DID.

Extension 1000 have also the from-internal context, but call don’t forward to extension. The log looks like:

pbx.c: Executing [s@from-internal:1] Goto(“Mobile/Nokla21-e58d”, “11,1”) in new stack
pbx_builtins.c: Goto (from-internal,11,1)
pbx.c: Executing [11@from-internal:1] ResetCDR(“Mobile/Nokla21-e58d”, “”) in new stack
pbx.c: Executing [11@from-internal:2] NoCDR(“Mobile/Nokla21-e58d”, “”) in new stack
pbx.c: Executing [11@from-internal:3] Progress(“Mobile/Nokla21-e58d”, “”) in new stack
pbx.c: Executing [11@from-internal:4] Wait(“Mobile/Nokla21-e58d”, “1”) in new stack
pbx.c: Executing [11@from-internal:5] Playback(“Mobile/Nokla21-e58d”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
file.c: <Mobile/Nokla21-e58d> Playing ‘silence/1.slin’ (language ‘en’)
file.c: <Mobile/Nokla21-e58d> Playing ‘cannot-complete-as-dialed.slin’ (language ‘en’)
file.c: <Mobile/Nokla21-e58d> Playing ‘check-number-dial-again.slin’ (language ‘en’)
pbx.c: Executing [11@from-internal:6] Wait(“Mobile/Nokla21-e58d”, “1”) in new stack
pbx.c: Executing [11@from-internal:7] Congestion(“Mobile/Nokla21-e58d”, “20”) in new stack
pbx.c: Spawn extension (from-internal, 11, 7) exited non-zero on ‘Mobile/Nokla21-e58d’
pbx.c: Executing [h@from-internal:1] Macro(“Mobile/Nokla21-e58d”, “hangupcall”) in new stack
pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“Mobile/Nokla21-e58d”, “1?theend”) in new stack
pbx_builtins.c: Goto (macro-hangupcall,s,3)
pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“Mobile/Nokla21-e58d”, “0?Set(CDR(recordingfile)=)”) in new stack
pbx.c: Executing [s@macro-hangupcall:4] NoOp(“Mobile/Nokla21-e58d”, " montior file= ") in new stack
pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“Mobile/Nokla21-e58d”, “1?skipagi”) in new stack
pbx_builtins.c: Goto (macro-hangupcall,s,7)
pbx.c: Executing [s@macro-hangupcall:7] Hangup(“Mobile/Nokla21-e58d”, “”) in new stack
app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘Mobile/Nokla21-e58d’ in macro ‘hangupcall’
pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘Mobile/Nokla21-e58d’

What make i wrong?

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@AndyInNYC wrote:

I’m upgrading from an earlier version of freepbx.
I have enabled faxing in User Manager
I have set a Trunk’s destination to ‘Fax Recipient’ and set it to virtual extension 6999
When I edit extension 6999 there is nowhere to add an email for fax nor to suggest that the default behavior is to answer a fax.

I have set an email in the User Manager for this extension, but is that all i need to do? That doesn’t seem right.

What steps do I need to take to get the trunk to send to the virtual extension and have the fax converted to pdf and sent to a specific email for that extension.

I can’t use Avantfax for this since I can’t get DID routing to work.

Thanks.

Andrew

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@thepossum905 wrote:

I have a fairly new freepbx 14 installation, the phones are on a remote network from the hosted pbx. Generally things are running swimmingly, although we’ve recently just enabled g729 in order to save on bandwidth.

There are some intermittent network issues with the internet provider, causing occasional call quality issues. All the boring stuff yup, a bit of extra packet loss and jitter that we can’t predict or prevent.
So we need to somehow visualize the per-call quality, in some sort of automated manner.

From the asterisk CLI, I can manually “pjsip show channelstats” and see live info on current active calls. Or I can script a “asterisk -rx ‘pjsip show channelstats’” to see everything at a periodic interval.

But what would be much more helpful would be for asterisk to log the per-call stats at the end of each call.

My google fu repeatedly brings up a command “rtcp set stats on”, but … nothing changes in the logs. How can I see the produced jitter data so that I can start to visualize it?

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@FreerPBXer wrote:

I’m reposting this from a locked thread because it didn’t get an answer.

I’ve read both these articles, just to preface.

https://wiki.freepbx.org/display/FPG/Time+Groups+Sample+Configurations
https://wiki.freepbx.org/display/FPG/Time+Group+User+Guide

In order to ease holiday scheduling we switched to defining non-working hours. And the two articles don’t say what happens if the start time for a time condition is after the end time (i.e. start at 5:00 PM and end at 8:00 AM). And if the end day for such a condition is Thursday, does it end with the change from positive match to negative at 8:00 AM, or from negative to positive at 5:00 PM? Similarly, with Monday would it start with a neg match at 8:00 AM or with a positive at 17:00?

If we need to define separately as 00:00 to 08:00 and also 17:00 to 23:59, then it seems like there would be a one minute unmatched state every night from 23:59 to 00:00.

Lastly, what if there is a conflict? i.e. we could define 08:00 to 17:00 M-F as work hours and set call flow based on that. But if we define a Monday as a holiday and designate after-hours call flow, which group ‘wins’?

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@bksales wrote:

Whats going on here? Is this a NAT issue?

They’re exactly 2 minutes apart each time. The SIP settings are the same default settings most PBXs have and x103 registers on 5060 like it’s supposed to. All are the same model phone (old Polycom 320) and have configs built from the same template.

-- Registered SIP '104' at x.x.x.x:4402
-- Registered SIP '102' at x.x.x.x:4403
-- Registered SIP '104' at x.x.x.x:4404
-- Registered SIP '102' at x.x.x.x:4405
-- Registered SIP '104' at x.x.x.x:4406
-- Registered SIP '102' at x.x.x.x:4407
-- Registered SIP '104' at x.x.x.x:4408
-- Registered SIP '102' at x.x.x.x:4409
-- Registered SIP '104' at x.x.x.x:4410
-- Registered SIP '102' at x.x.x.x:4411
-- Registered SIP '104' at x.x.x.x:4412
-- Registered SIP '102' at x.x.x.x:4413
-- Registered SIP '104' at x.x.x.x:4414
-- Registered SIP '102' at x.x.x.x:4415
-- Registered SIP '104' at x.x.x.x:4416

SIP settings

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@thepossum905 wrote:

I have a fairly new freepbx 14 installation, the phones are on a remote network from the hosted pbx. Generally things are running swimmingly, although we’ve recently just enabled g729 in order to save on bandwidth.

There are some intermittent network issues with the internet provider, causing occasional call quality issues. All the boring stuff yup, a bit of extra packet loss and jitter that we can’t predict or prevent.
So we need to somehow visualize the per-call quality, in some sort of automated manner.

From the asterisk CLI, I can manually “pjsip show channelstats” and see live info on current active calls. Or I can script a “asterisk -rx ‘pjsip show channelstats’” to see everything at a periodic interval.

But what would be much more helpful would be for asterisk to log the per-call stats at the end of each call.

My google fu repeatedly brings up a command “rtcp set stats on”, but … nothing changes in the logs. How can I see the produced jitter data so that I can start to visualize it?

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@dux wrote:

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.

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@thepossum905 wrote:

I have a fairly new freepbx 14 installation, the phones are on a remote network from the hosted pbx. Generally things are running swimmingly, although we’ve recently just enabled g729 in order to save on bandwidth.

There are some intermittent network issues with the internet provider, causing occasional call quality issues. All the boring stuff yup, a bit of extra packet loss and jitter that we can’t predict or prevent.
So we need to somehow visualize the per-call quality, in some sort of automated manner.

From the asterisk CLI, I can manually “pjsip show channelstats” and see live info on current active calls. Or I can script a “asterisk -rx ‘pjsip show channelstats’” to see everything at a periodic interval.

But what would be much more helpful would be for asterisk to log the per-call stats at the end of each call.

My google fu repeatedly brings up a command “rtcp set stats on”, but … nothing changes in the logs. How can I see the produced jitter data so that I can start to visualize it?

Posts: 16

Participants: 4

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