Re: [asterisk-users] Asterisk 13 High CPU usage

[ian gilmour]

The following bash 1-liner may be useful...

while true; do top -Hbc -p `pgrep asterisk` -n 1 && asterisk -rx "core show
threads"; sleep 1; done

Regards,

Ian

On 24/07/2016 13:39, Tzafrir Cohen wrote:

>

> On Fri, Jul 22, 2016 at 12:02:43AM +0100, Chirag Desai wrote:


>>
>> I am not sure where to start looking in order to debug the CPU usage by
asterisk and would very much appreciate some guidance.
>
> If you run 'top', the basic information would be to show per-CPU
information (press '1'). Another thing to look at: press 'H' to get
per-thread entries. Do you have many many threads each taking a small part
of a core, or a few threads taking lots of CPU time? I believe that the PID
(process/thread ID) you see in top is also the second item in each line in
the output of 'core show threads'. So this could give you some clues
regarding the CPU hogs you see in top.
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[asterisk-users] max concurrent calls with bundled pjproject

[ian gilmour]

Hi,

PJSIP in the past had limitations on the max concurrent calls, etc. There were 
ways to overcome them by changing the source code. (e.g. 
http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2013-February/015721.html
 
)

Do any similar tweaks need to be done to the bundled pjproject to handle high 
volumes of concurrent calls with Asterisk?

What (if any) are the current default asterisk 13 + pjproject audio + video 
concurrent call limits if using the bundled pjproject + asterisk patches as is?

Thanks in advance.

Regards,

IanG-- 
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Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

[Ian Gilmour]

Hi,

I don’t see any SIP ACK’s in your trace.

Is the SIP 200 OK reaching the originating caller, or being blocked on 
the way through?


Asterisk will tear down the call after ~30secs of audio playing in both 
directions if it doesn't receive the SIP ACK.


Regards,

Ian


On 15/10/2016 12:05, Andre Gronwald wrote:

hi,
let me explain in detail, what i have configured and what is happening 
now:


1st router w724v (Deutsche Telekom AG):
- port forwarding, everything to destination port 51000-55999 to 
device with ip 192.168.2.50 (interface of 2nd router)

2nd router Bintec RS353j):
- configured NAT, everything to port 51000-55999 to device 
192.168.3.99 (same ports)


other direction is totally open.

I observed that all sip calls are closed exactly after 32s. call is 
disconnected on calling side as well... seems to be a timeout issue.


here i have some debug logs. I see lot of requests from asterisk to 
sipgate.de, which are not answered. but communication is going fine in 
both directions (otherwise registration would not be possible?):



<--- Received SIP request (1302 bytes) from UDP:217.10.79.9:5060 --->
INVITE sip:2636146e0@80.142.13.32:55060 SIP/2.0
Via: SIP/2.0/UDP 
217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0

Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: 
Record-Route: 
Record-Route: 
From: "02363361779" ;tag=as02fa8fcc
To: 
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
CSeq: 103 INVITE
Contact: 
max-forwards: 66
supported: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE

Content-Type: application/sdp
Content-Length:   394

v=0
o=root 15363811 15363812 IN IP4 192.168.2.1
s=sipgate VoIP GW
c=IN IP4 192.168.2.1
t=0 0
m=audio 7070 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<--- Transmitting SIP response (733 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0

Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: 
Record-Route: 
Record-Route: 
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
From: "09" ;tag=as02fa8fcc
To: 
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Content-Length:  0

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0

Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: 
Record-Route: 
Record-Route: 
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
From: "09" ;tag=as02fa8fcc
To: ;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: 
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE

Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (1302 bytes) from UDP:217.10.79.9:5060 --->
INVITE sip:2636146e0@80.142.13.32:55060 SIP/2.0
Via: SIP/2.0/UDP 
217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0

Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: 

Re: [asterisk-users] I think this is a bug (video call file) 11.25.1 and 13.13.1

[ian gilmour]

Hi Jerry,

You don't say how you created the video file?

Asterisk video support is v. basic. It can only playback video it created
using it's own specific file format for the video file. In general you
cannot use it to playback videos created by 3rd party tools.

What does work (if you have asterisk configured correctly)

Invoke Asterisk Record() from a video capable sip client (e.g.
linphone/jitsi should work fine here) and which is configured to support
audio Make a call. Asterisk will create the audio file and an
accompanying video file  (e.g. .h264) in the same directory. Now invoke
Asterisk Playback() to play the files back. If the SIP client and Asterisk
negotiate the same video codec that was used to record the initial call
then Asterisk will play back both audio+video streams.

Note: Asterisk does no video transcoding.

Note: If video plays back at the wrong speed it's possibly a result of
https://issues.asterisk.org/jira/browse/ASTERISK-26554.

Regards,

Ian
On 21/12/2016 00:01, Jerry Geis wrote:

>Hi Jerry,
> just had a look through the code, and from what I can tell, what
>you're trying to do is not supposed to work, exactly. It appears that
>what Asterisk expects is to be given a filename, such as "myplayback".
>Asterisk will first search for an audio version of the file (like
>myplayback.gsm or myplayback.opus), and open that as an audio stream. If
>that succeeds, it then will also see if there is an accompanying video
>stream (such as myplayback.h264). If it then finds that video, then the
>result will be that Asterisk will play the audio from the audio file and
>the video from the video file.

>What this means is that Asterisk does not properly handle:
>* Files that have audio and video streams contained within
>* Video files without accompanying audio

>This is one of those times where Asterisk's handling of video is not
>user-friendly and in general ass-backwards and terrible. If you have a
>tool that can extract the audio to its own file, then you would be able
>to run your scenario, presumably.

>It would be a welcome addition for Asterisk to be able to open a single
>file containing video and accompanying audio and be able to play those back.

Hi Mark,

Thanks for your reply...
I just tried what you suggested on only got audio. I created a wav file and
put it in the /tmp
directory just like the video.h264 file. So /tmp has video.h264 and
video.wav both.
I then placed the call and only heard the audio from the wav file.

I used this for my call file:
Channel: SIP/2002
Context: testing
Extension: 99
Priority: 1
Application: Playback
Codecs: h263,h264,vp8,g722,ulaw,alaw,wav
Data: /tmp/video

My Bria 4 softphone uses the h263 and h264 codecs and of course wav file
audio.
Based on your look of the code did I miss something to trigger the playing
of the video file?
I can extract the audio out to a seperate file - so not a show stopper for
me.

No errors showed up on the Asterisk CLI when I did my test.

Thanks so much,

Jerry
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Re: [asterisk-users] Confbridge GUI?

[Ian Gilmour]

Hi,

> On 17 Oct 2017, at 18:30, Richard Kenner  wrote:
> 
>> If you can provide details, even vague ones, about how you did it, I
>> can update the WMM package.
> 
> See http://asterisk.gnat.com/meetme.tgz 
> 
> That's a gzipped tar of our working directory plus the relevant parts of
> extensions.conf.  I xxx'ed out phone numbers and Google interface data.

The above tarball appears to be no longer available.

Does anyone have a copy they can put up somewhere public?

Thanks in advance,

Ian

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