Hallo
Ich hab hier ein Problem mit dem Verbindungsaufbau.
Habe laut Anleitung eine kleine Konfiguration gemacht:
/etc/asterisk/sip.conf
/etc/asterisk/extensions.conf
Die Linux-Box und der PC mit dem X-Lite phone sind über einen Switch
sozusagen direkt verbunden.
Beim start von Asterisk kommt dies Meldung
X-Lite kann sich aber nicht anmelden:
Weis jemand weiter?
Ich hab hier ein Problem mit dem Verbindungsaufbau.
Habe laut Anleitung eine kleine Konfiguration gemacht:
/etc/asterisk/sip.conf
Code:
; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn't, try adding "nat=1" to each peer definition to
; solve translation problems.
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
[1000]
username=1000
secret=abc123
context=mytest
host=dynamic
/etc/asterisk/extensions.conf
Code:
[general]
static=yes
writeprotect=yes
;Suport phones
;SUPPORTPHONES=SIP/2205&SIP/2206&SIP/2207&SIP/2208&SIP/2209
[globals]
XLITE=SIP/1000
[mytest]
exten => 1367,1Dial(SIP/1000)
exten => 2890,1,Wait(2)
exten => 2890,2,Answer
exten => 2890,3,Playback(demo-echotest)
exten => 2890,4,Echo()
exten => 2468,1,Dial($XLITE)
Die Linux-Box und der PC mit dem X-Lite phone sind über einen Switch
sozusagen direkt verbunden.
Beim start von Asterisk kommt dies Meldung
Code:
[pbx_config.so] => (Text Extension Configuration)
== Parsing '/etc/asterisk/extensions.conf': Found
== Setting global variable 'XLITE' to 'SIP/1000'
-- Registered extension context 'mytest'
-- Added extension '1367' priority 1 to mytest
-- Added extension '2890' priority 1 to mytest
-- Added extension '2890' priority 2 to mytest
-- Added extension '2890' priority 3 to mytest
-- Added extension '2890' priority 4 to mytest
-- Added extension '2468' priority 1 to mytest
X-Lite kann sich aber nicht anmelden:
Code:
SEND TIME: 10157385
SEND >> 10.100.249.12:5060
REGISTER sip:10.100.249.12 SIP/2.0
Via: SIP/2.0/UDP 10.100.249.86:5060;rport;branch=z9hG4bKFAC1B6F2B5414EE9855696A09A83FB22
From: Tester <sip:[email protected]>;tag=3354744682
To: Tester <sip:[email protected]>
Contact: "Tester" <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 4806 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0
RECEIVE TIME: 10157385
RECEIVE << 10.100.249.12:5060
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.100.249.86:5060;branch=z9hG4bKFAC1B6F2B5414EE9855696A09A83FB22
From: Tester <sip:[email protected]>;tag=3354744682
To: Tester <sip:[email protected]>;tag=as7ae925e2
Call-ID: [email protected]
CSeq: 4806 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0
Weis jemand weiter?