Есть небольшой прогресс! (думаю что осталось немного)
Телефон avaya 9620 прошивка SIP96xx_2_2_0_0.bin
установлены ip вручную:
ip телефона 192.168.2.46
ip http сервер 192.168.2.103 (он же и астериск)
Файл 46xxsettings.txt:
SET SIPPROXYSRVR "192.168.2.103"
SET SIPSIGNAL 0
SET ENABLE_AVAYA_ENVIRONMENT 0
Телефон стартует, забирает файл с настройками 46xxsettings.txt с веб-сервера и пытается зарегаться на астериске, но не может с ним договориться
на телефоне просто висит надпись Logging in...
На астериске sip.conf:
[general]
context=default ; Default context for incoming calls
allowoverlap=yes ; Disable overlap dialing support. (Default is yes)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=no ; Enable DNS SRV lookups on outbound calls
dtmfmode=auto
canreinvite=no
useragent="pngh router"
rfc2833compensate=yes
jbenable=yes
jbmaxsize=180
tcpenable=yes
[authentication]
[1001]
port=5060
type=friend
host=dynamic
username=1001
canreinvite=no
context=home
callerid="avaya9620"
disallow=all
allow=alaw
allow=ulaw
dtmfmode=auto
Когда телефон пытается зарегаться на asteriske картина следующая:
home-asterisk*CLI> sip set debug
SIP Debugging re-enabled
home-asterisk*CLI>
<--- SIP read from 192.168.2.46:5060 --->
REGISTER sip:192.168.2.103 SIP/2.0
From: sip:1001@192.168.2.103;tag=-1b50f386d43fb0_F192.168.2.46
To: sip:1001@192.168.2.103
Call-ID: 1_309011d11b8e386d74a0_R@192.168.2.46
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 192.168.2.46;branch=z9hG4bK1_30904615c35c386d74b2_R
Content-Length: 0
Max-Forwards: 70
Contact: <sip:1001@192.168.2.46;transport=udp>;q=1;expires=3600
Allow: INVITE,CANCEL,BYE,ACK,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE
User-Agent: Avaya one-X Deskphone
upported: replaces
<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.2.46 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.2.46:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.46;branch=z9hG4bK1_30904615c35c386d74b2_R;received=192.168.2.46
From: sip:1001@192.168.2.103;tag=-1b50f386d43fb0_F192.168.2.46
To: sip:1001@192.168.2.103
Call-ID: 1_309011d11b8e386d74a0_R@192.168.2.46
CSeq: 1 REGISTER
User-Agent: "pngh router"
llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 192.168.2.46:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.46;branch=z9hG4bK1_30904615c35c386d74b2_R;received=192.168.2.46
From: sip:1001@192.168.2.103;tag=-1b50f386d43fb0_F192.168.2.46
To: sip:1001@192.168.2.103;tag=as49fd597e
Call-ID: 1_309011d11b8e386d74a0_R@192.168.2.46
CSeq: 1 REGISTER
User-Agent: "pngh router"
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:1001@192.168.2.46;transport=udp>;expires=3600
Date: Sun, 21 Feb 2010 09:53:59 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1_309011d11b8e386d74a0_R@192.168.2.46' in 32000 ms (Method: REGISTER)
home-asterisk*CLI>
<--- SIP read from 192.168.2.46:5060 --->
REGISTER sip:192.168.2.103 SIP/2.0
From: sip:1001@192.168.2.103;tag=-1b50f386d43fb0_F192.168.2.46
To: sip:1001@192.168.2.103
Call-ID: 1_309011d11b8e386d74a0_R@192.168.2.46
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 192.168.2.46;branch=z9hG4bK1_30904615c35c386d74b2_R
Content-Length: 0
Max-Forwards: 70
Contact: <sip:1001@192.168.2.46;transport=udp>;q=1;expires=3600
Allow: INVITE,CANCEL,BYE,ACK,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE
User-Agent: Avaya one-X Deskphone
upported: replaces
<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.2.46 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.2.46:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.46;branch=z9hG4bK1_30904615c35c386d74b2_R;received=192.168.2.46
From: sip:1001@192.168.2.103;tag=-1b50f386d43fb0_F192.168.2.46
To: sip:1001@192.168.2.103
Call-ID: 1_309011d11b8e386d74a0_R@192.168.2.46
CSeq: 1 REGISTER
User-Agent: "pngh router"
llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
home-asterisk*CLI>
<--- Transmitting (no NAT) to 192.168.2.46:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.46;branch=z9hG4bK1_30904615c35c386d74b2_R;received=192.168.2.46
From: sip:1001@192.168.2.103;tag=-1b50f386d43fb0_F192.168.2.46
To: sip:1001@192.168.2.103;tag=as49fd597e
Call-ID: 1_309011d11b8e386d74a0_R@192.168.2.46
CSeq: 1 REGISTER
User-Agent: "pngh router"
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:1001@192.168.2.46;transport=udp>;expires=3600
Date: Sun, 21 Feb 2010 09:54:05 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1_309011d11b8e386d74a0_R@192.168.2.46' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog '1_6f9-44e72345386d4a9b_R@192.168.2.46' Method: REGISTER
Really destroying SIP dialog '1_309011d11b8e386d74a0_R@192.168.2.46' Method: REGISTER
Подскажите пожалуйста - где копать?