packet retransmission / lost packet

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packet retransmission / lost packet

Postby koceila bauer » Mon May 24, 2021 3:07 pm

hello guys , im new in this forum and its my fisrt post , please i have a problem and i have not found solution after 15 days of search. , i dont have problem to log carrier , all is okey , but when i put the local ip adresse of server (192.168.1.230) in the dmz of router to connect from remote , the softphone dont register (from remote ) , and there are no voice when agent page call softphone (in local )

i have installed vicibox 9.0.3 in a dell server with vm ware esxi 6.5 and linux opensuse 64 bits

i got this message in putty in asterisk -rvvc (the softphone passeword


[May 24 19:41:40] NOTICE[2622]: chan_sip.c:28751 handle_request_register: Registration from '<sip:2000@41.220.144.71>' failed for '144.217.73.74:53605' - Wrong password
[May 24 19:41:41] WARNING[2622]: chan_sip.c:4128 retrans_pkt: Timeout on d935fd231ebb85979b26573d0ec69dc4 on non-critical invite transaction.
[May 24 19:41:57] NOTICE[2622]: chan_sip.c:28751 handle_request_register: Registration from '<sip:24@41.220.144.71>' failed for '144.217.73.74:56155' - Wrong password
[May 24 19:42:02] == Manager 'sendcron' logged on from 127.0.0.1
[May 24 19:42:02] == Manager 'sendcron' logged off from 127.0.0.1
[May 24 19:42:02] == Manager 'sendcron' logged on from 127.0.0.1
[May 24 19:42:03] NOTICE[2622]: chan_sip.c:28751 handle_request_register: Registration from '<sip:8925@41.220.144.71>' failed for '144.217.73.74:56920' - Wrong password
[May 24 19:42:04] == Manager 'sendcron' logged off from 127.0.0.1
[May 24 19:42:07] == Manager 'sendcron' logged on from 127.0.0.1
[May 24 19:42:07] == Manager 'sendcron' logged off from 127.0.0.1
[May 24 19:42:13] NOTICE[2622]: chan_sip.c:28751 handle_request_register: Registration from '<sip:495@41.220.144.71>' failed for '144.217.73.74:58351' - Wrong password
[May 24 19:42:18] WARNING[2622]: chan_sip.c:4128 retrans_pkt: Timeout on 889859988-2101649084-1680259874 on non-critical invite transaction.
[May 24 19:42:21] WARNING[2622]: chan_sip.c:4128 retrans_pkt: Timeout on 1433880403-641930140-1057249242 on non-critical invite transaction.
[May 24 19:42:25] WARNING[2622]: chan_sip.c:4128 retrans_pkt: Timeout on 2124166709-599190679-783336406 on non-critical invite transaction.
[May 24 19:42:26] WARNING[2622]: chan_sip.c:4128 retrans_pkt: Timeout on a1baaf935ca38c178211e78043bf3cf5 on non-critical invite transaction.


sip.conf :

[general]
context=trunkinbound ; Default context for incoming calls
allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
;realm=mydomain.tld ; Realm for digest authentication
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;domain=mydomain.tld ; Set default domain for this host
;pedantic=yes ; Enable checking of tags in headers,
;tos_sip=cs3 ; Sets TOS for SIP packets.
;tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
;maxexpiry=3600 ; Maximum allowed time of incoming registrations
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=gsm
mohinterpret=default
mohsuggest=default
language=en ; Default language setting for all users/peers
relaxdtmf=yes ; Relax dtmf handling
trustrpid = no ; If Remote-Party-ID should be trusted
sendrpid = yes ; If Remote-Party-ID should be sent
progressinband=no ; If we should generate in-band ringing always
;useragent=Asterisk PBX ; Allows you to change the user agent string
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
;compactheaders = yes ; send compact sip headers.
videosupport=no ; Turn on support for SIP video. You need to turn this on
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
callevents=yes ; generate manager events when sip ua
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
;regcontext=sipregistrations
rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
;sipdebug = yes ; Turn on SIP debugging by default, from
;recordhistory=yes ; Record SIP history by default
;dumphistory=yes ; Dump SIP history at end of SIP dialogue
;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
limitonpeers = yes ; Apply call limits on peers only. This will improve
;t38pt_udptl = yes ; Default false
;register => 1234:password@mysipprovider.com
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
externip = 197.113.187.124 ; Address that we're going to put in outbound SIP
;externhost=test.test.com ; Alternatively you can specify a domain
;externrefresh=10 ; How often to refresh externhost if
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
nat=yes ; Global NAT settings (Affects all peers and users)
canreinvite=no ; Asterisk by default tries to redirect the
;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
;rtsavesysname=yes ; Save systemname in realtime database at registration
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
;ignoreregexpire=yes ; Enabling this setting has two functions:
;domain=mydomain.tld,mydomain-incoming
;domain=1.2.3.4 ; Add IP address as local domain
;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
;autodomain=yes ; Turn this on to have Asterisk add local host
;fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize = 100 ; Max length of the jitterbuffer in milliseconds.
jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
qualify=yes ; By default, qualify all peers at 2000ms
limitonpeer = yes ; enable call limit on a per peer basis, different from limitonpeers

#include sip-vicidial.conf

; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:test@10.10.10.16:5060
;
; setup account for SIP trunking:
; [SIPtrunk]
; disallow=all
; allow=ulaw
; allow=alaw
; type=friend
; username=testSIPtrunk
; secret=test
; host=10.10.10.16
; dtmfmode=inband
; qualify=1000


sip.vicidial conf


[FRANCE]
disallow=all
allow=alaw
allow=g729
type=friend
username=****
secret=******
host=51.********
dtmfmode=rfc2833
context=trunkinbound
sendrpid=yes
nat=yes
insecure=port,invite
qualify=yes
Last edited by koceila bauer on Tue May 25, 2021 12:30 am, edited 1 time in total.
koceila bauer
 
Posts: 3
Joined: Mon May 24, 2021 10:24 am

Re: packet retransmission / lost packet

Postby carpenox » Mon May 24, 2021 4:51 pm

your server is being targetted by hackers, please secure it using this article from my blog: https://cyburityllc.com/?p=1977
Leap 15.3 | Version: 2.14-833a | BUILD: 211106-1500 | SVN Version: 3542 | DB Schema Version: 1648 | Asterisk 16.20.0-vici
www.CyburDial.net -:- 844-PC-SATA-2 -:- My Blog: http://vicidial.blog -:- Whatsapp: +19549477572 -:- Skype: live:carpenox_3
carpenox
 
Posts: 1707
Joined: Wed Apr 08, 2020 2:02 am
Location: Coral Springs, FL

Re: packet retransmission / lost packet

Postby koceila bauer » Tue May 25, 2021 12:18 am

hello. , thank you so much for your reply , please where is the problem in my server ? its just because i havent secure it ??
koceila bauer
 
Posts: 3
Joined: Mon May 24, 2021 10:24 am


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