como conectar um softphone ao asterisco

Question

Para contextualizar, aqui está minha rede:

Noservidortleilax,háclaramenteumresultadodesipsaknodoge:

tleilax*CLI>tleilax*CLI>coreshowversionAsterisk1.8.29.0-vicibuiltbyabuild@cloud110onax86_64runningLinuxon2014-08-2123:18:17UTCtleilax*CLI>[Feb2021:06:19]<---SIPreadfromUDP:192.168.1.3:44226--->OPTIONSsip:345@tleilaxSIP/2.0Via:SIP/2.0/UDP127.0.1.1:44226;branch=z9hG4bK.508a6d72;rport;aliasFrom:sip:[email protected]:44226;tag=2a099edcTo:sip:345@tleilaxCall-ID:[email protected]:1OPTIONSContact:sip:[email protected]:44226Content-Length:0Max-Forwards:0User-Agent:sipsak0.9.6Accept:text/plain<------------->[Feb2021:06:19]---(11headers0lines)---[Feb2021:06:19]Lookingfor345intrunkinbound(domaintleilax)[Feb2021:06:19]<---Transmitting(NAT)to192.168.1.3:44226--->SIP/2.0200OKVia:SIP/2.0/UDP127.0.1.1:44226;branch=z9hG4bK.508a6d72;alias;received=192.168.1.3;rport=44226From:sip:[email protected]:44226;tag=2a099edcTo:sip:345@tleilax;tag=as5d21da5cCall-ID:[email protected]:1OPTIONSServer:AsteriskPBX1.8.29.0-viciAllow:INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGESupported:replaces,timerContact:<sip:192.168.1.2:5060>Accept:application/sdpContent-Length:0<------------>[Feb2021:06:19]SchedulingdestructionofSIPdialog'[email protected]'in32000ms(Method:OPTIONS)[Feb2021:06:38]ReallydestroyingSIPdialog'[email protected]'Method:OPTIONS[Feb2021:06:51]ReallydestroyingSIPdialog'[email protected]'Method:OPTIONStleilax*CLI>exittleilax:~#

Emdoge,enviandoamensagemsipsak:

thufir@doge:~$thufir@doge:~$sudosipsak-vv-ssip:345@tleilax-m"hi"
No SRV record: _sip._tcp.tleilax
No SRV record: _sip._udp.tleilax
using A record: tleilax
Max-Forwards set to 0

message received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.1.1:44226;branch=z9hG4bK.508a6d72;alias;received=192.168.1.3;rport=44226
From: sip:[email protected]:44226;tag=2a099edc
To: sip:345@tleilax;tag=as5d21da5c
Call-ID: [email protected]
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.29.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.1.2:5060>
Accept: application/sdp
Content-Length: 0



** reply received after 0.794 ms **
   SIP/2.0 200 OK
   final received
thufir@doge:~$ 

Eu tentei alguns telefones, isso é sflphone conectando:

This assistant is now finished.
You can at any time check your registration state or modify your accounts parameters in the Options/Accounts window.

Alias :   345
Server :   tleilax
Username :   345
Security: None

O que não indica uma mensagem "200 OK" indicando que está conectado. Como faço para solucionar problemas de conectividade? Esses dois computadores estão na mesma rede, podem pingar uns aos outros ou até ssh ; Eu não entendo porque o softphone em doge tem problemas em receber uma mensagem "200 OK" de tleilax - apenas diz "tentando", e outros softphones dão resultados similares.

os usuários:

tleilax*CLI> 
tleilax*CLI> sip show users
Username                   Secret           Accountcode      Def.Context      ACL  Forcerport
101                        password         101              default          No   Yes       
gs102                      password         gs102            default          No   Yes       
tleilax*CLI> 
tleilax*CLI> sip show user 101


  * Name       : 101
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : default
  Language     : en
  Accountcode  : 101
  AMA flags    : Unknown
  Netborder CPD: No
  Transfer mode: open
  MaxCallBR    : 384 kbps
  CallingPres  : Presentation Allowed, Not Screened
  Call limit   : 0
  Callgroup    : 
  Pickupgroup  : 
  Callerid     : "" <101>
  ACL          : No
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Sess-Min-SE  : 90 secs
  RTP Engine   : asterisk
  Codec Order  : (ulaw:20,gsm:20)
  Auto-Framing:  No 

tleilax*CLI> 

arquivos de configuração:

tleilax:~ # 
tleilax:~ # cat /etc/asterisk/sip.conf
[general]

...

#include sip-vicidial.conf

; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:[email protected]: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
tleilax:~ # 
tleilax:~ # cat /etc/asterisk/sip-vicidial.conf
; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST



[101]
username=101
secret=password
accountcode=101
callerid="" <101>
mailbox=101
context=default
type=friend
host=dynamic

[gs102]
username=gs102
secret=password
accountcode=gs102
callerid="Test Admin Phone" <>
mailbox=102
context=default
type=friend
host=dynamic


; END OF FILE    Last Forced System Reload: 2015-02-20 16:49:28
tleilax:~ # 

    

Answer 1

Você precisa configurar o localnet e o nat de acordo

externip = X.X.X.X
fromdomain = yourdomain.com
localnet = 192.168.X.0/255.255.255.0
qualify=yes

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