Hi,
Is it still necessary to care about the minimal free memory amount in the
TLS module configuration (e.g., the low_mem_threshold1/2 parameters) in
Ubuntu 18.04 with openssl v.1.1.1?
I tried to find some more information about the "openssl bug #1491" but the
1491 issue in the current Openssl bug tracking system seems to be not
related to the memory problem:
https://github.com/openssl/openssl/issues/1491
Best regards,
Leonid Fainshtein
Xorcom Ltd
I have a setup where we wanted to just build a android app linphone to SIP-PBX.
But come to find out upon further research due to potential battery drain/background app usage. The device isn’t registered all the time and need some sort of push notification to google firebase back to android app to re-setup the register / INVITE call.
Is there a module/config of this kind of setup. (now could I use cloud kamailio or would this have to be onsite local to the PBX).
Hello!
We have a next call-flow topology: User A -> FreeSWITCH -> Kamailio Fork ->
Kamailio Edge => User(s) B, C, D, ...
- FreeSWITCH adds X- header with a list of users to fork call (B, C, D
legs).
- Kamailio Fork generates branches via append_branch function towards
Kamailio Edge.
- Kamailio Edge keeps customer registration and sends BLF statuses
(pua_dialoginfo).
The problem (or normal behavior by RFC) is that the dialog module on
Kamailio Edge does not split these incoming legs into separated dialogs
(incoming SIP Call-ID and from tag are the same).
And as per my checking *kamcmd dlg.list_ctx* shows only the structure from
the first leg and finally, it leads to a problem with BLF (as pua_diloginfo
loses some SIP PUBLISHes).
Any advice on how to fix this is appreciated!
--
BR,
Denys Pozniak
Dear Team,
I am using kamailio with rtpengine for media.
When I make call between two sipUA then media flow is manage by rtpengine
offer/answer.
The issue is when hold the call then following some error are coming .
Pls find the full log in attached file.
rtpengine[14099]: DEBUG:
[43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: av_log:
parser not found for codec pcm_s16le, packets or times may be invalid.
rtpengine[14099]: DEBUG:
[43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: av_log:
max_analyze_duration 5000000 reached at 5120000 microseconds st:0
rtpengine[14099]: DEBUG:
[43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: av_log:
stream 0: start_time: -1152921504606847.000 duration: 27.034
rtpengine[14099]: DEBUG:
[43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: av_log:
format: start_time: -9223372036854.775 duration: 27.034 bitrate=128 kb/s
rtpengine[14099]: DEBUG:
[43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: av_log:
After avformat_find_stream_info() pos: 131130 bytes read:196666 seeks:1
frames:22
rtpengine[14099]: ERR:
[43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: No supported
output codec found in SDP
rtpengine[14099]: INFO:
[43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: Replying to
'play media' from 127.0.0.1:48757 (elapsed time 0.121111 sec)
rtpengine[14099]: DEBUG:
[43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: Response
dump for 'play media' to 127.0.0.1:48757: { "result": "ok" }
I am unable to understand this issue.
Is this related to rtpenging, which is not supported given codec or file
codec, which is used to play after hold call or anything else in script
issue.
Can you please guide me where is the issue.
Regards
Amit Pal
I use this construction to check the presence of path in device registration
if(reg_fetch_contacts("location", "$fu", "caller") &&
$(ulc(caller=>path)[0]) == $null) {
handle_ruri_alias();
}
Is possible to check path presence after lookup("location") function call?
Sergey
Dear Team,
I am using kamailio with rtpengine for media.
when call getting hold no media sound play (configure hold.mp3).
>From rtpengine gets following error.
rtpengine[624]: ERR: No suitable SDP section for media playback
rtpengine[624]: WARNING: Protocol error in packet from 127.0.0.1:46333:
Failed to start media playback from file [d8:supportsl10:load
limite4:file38:/home/coralswitch/queuesounds/hold.mp37:call-id60:89162207232
fae50MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.13:received-froml3:IP414:19
2.168.20.254e8:from-tag32:9f5288f14b704b28a1c2dfc2112250eb6:to-tag8:1e05e757
7:command10:play mediae]
rtpengine[624]: INFO: Received command 'offer' from 127.0.0.1:46333
rtpengine[624]: INFO: Replying to 'offer' from 127.0.0.1:46333 (elapsed time
0.000106 sec)
/usr/local/sbin/kamailio[9094]: ERROR:rtpengine [rtpengine.c:2619]:
rtpp_function_call(): proxy replied with error: Failed to start media
playback from file
Attached file show complete log of one call with hold /unhold actions.
Please advise what I miss to configure here.
Regards
Amit Pal
