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Logging jitter stats per-call

July 22, 2019, 1:39 pm
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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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How to read a Zulu changelog

October 21, 2019, 2:16 am
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@dan_ce wrote:

Really, really minor query here, but how does one see what Zulu upgrades have fixed? The changelog just says

**14.0.57.7:**  Packaging of ver 14.0.57.7
**14.0.57.6:**  Packaging of ver 14.0.57.6
**14.0.57.5:**  Packaging of ver 14.0.57.5
**14.0.57.4:**  Packaging of ver 14.0.57.4
**14.0.57.3:**  Packaging of ver 14.0.57.3
**14.0.57.2:**  Packaging of ver 14.0.57.2
**14.0.57.1:**  Packaging of ver 14.0.57.1

I was hoping the bug 926702 might have been fixed, relating to the problem that the caller name isn’t coming through on the windows desktop client.

(EDIT: Just did some testing, bug isn’t fixed as far as I can see)

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Logging jitter stats per-call

July 22, 2019, 1:39 pm
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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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Ok to delete var/spool/asterisk/tmp?

October 25, 2019, 1:53 pm
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@mvogel4949 wrote:

I have a system that is quite high in storage % and I couldn’t quite figure out why. I am recording calls but that only adds up to 61G out of the 178G being used. I went searching and found var/spool/asterisk/tmp that looks to be quite large. There are various wav files along with a backup-2 file that contains etc,tftp boot and var folders. Perhaps a backup that was stopped? Can I delete this tmp file?

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Logging jitter stats per-call

July 22, 2019, 1:39 pm
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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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Anybody using Magnus Billing?

October 29, 2019, 10:52 am
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@dcitelecom wrote:

Since a2billing is no longer supported I am trying to find alternate solutions. Magnusbilling looks interesting and I was wondering if anyone has tried it and can provide some feedback.

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Participants: 1

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Logging jitter stats per-call

July 22, 2019, 1:39 pm
$
0
0

@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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Asterisk 17

November 3, 2019, 9:02 pm
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@dux wrote:

Asterisk 17 has already been released. When will it be available in FreePBX installation repository?

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Logging jitter stats per-call

July 22, 2019, 1:39 pm
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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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SIP Proxy config

November 5, 2019, 8:34 pm
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@simpleman wrote:

Hi all. I’m newbie .I have a FreePBX 13 for SIP server and build a Asterisk 13 on Ubuntu 16.04 for SIP Proxy.
So, What is all I need to do in SIP Proxy? . I need some instruction
I think, I need config Trunk in sip.extension between FreePBX 13 and SIP proxy. Dial plan in extensions.conf.
Thanks for readding. Sorry about bad English .

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Participants: 2

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UnixODBC update

November 5, 2019, 11:59 pm
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@dux wrote:

Hello, everyone. I have a FreePBX distro. How can I upgrade unixODBC to version 2.3.7 with yum? The current version is 2.3.1 Yum does not show any upgrade in the repository.

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Participants: 1

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Understanding Lenny files

November 6, 2019, 12:07 am
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@nobby6 wrote:

Lenny as we all know and dearly love, repeats himself,… except Lenny1
So how do I introduce a Lenny0, along with Lenny 1 that wont get repeated?

I don’t understand the
,Set(i=${IF($[“0${i}”=“016”]?7:$[0${i}+1])})

Which no doubt is where it all happens, nor can I find any “good” references as to exactly what each of this is doing.

Any pointers?

(basically i have a ulaw file Lenny0 which has some local phone rings, and a couple of touch tone buttons pressed, that I want to insert before Lenny dopes all his magic, all other attempts like using an earlier Playback() fail dismally, thats if it doesnt get ignored or creash poor 'ol len altogether :slight_smile:

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Logging jitter stats per-call

July 22, 2019, 1:39 pm
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0
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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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Time Conditions, need clarifications when defining based on non-work hours

July 22, 2019, 8:35 pm
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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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Cdr not show in freepbx14 asterisk 13

July 23, 2019, 3:23 am
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LetsEncrypt failing to mirror1.freepbx.org:80

November 9, 2019, 2:03 pm
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Restore fails from v14 to v15

November 9, 2019, 2:06 pm
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@jameeldroid wrote:

hi, did fresh install v15 1910 and restored backup of config from v14 all the restore output went well with an error in the very end "the process “fwconsole extip” exceeded the timeout of 60 seconds. i closed the restore output window and clicked apply settings but it came up with unknown error and asked me to fwconsole reload --verbose… when i do this… it throws the same error on CLI too… "the process “fwconsole extip” exceeded the timeout of 60 seconds.

please help.

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Filestore , S3

November 9, 2019, 9:08 pm
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@psdk wrote:

Hi to all,

I know that it’s new feature, but anyone knows how S3 bucket will mount in the system? (Riofs, …)
Is there any experience to use it for file recording storage?

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Participants: 1

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Logging jitter stats per-call

July 22, 2019, 1:39 pm
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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

Read full topic

↧

Time Conditions, need clarifications when defining based on non-work hours

July 22, 2019, 8:35 pm
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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’?

Posts: 4

Participants: 3

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