Hello, i'm trying to get work mizu webphone with asterisk 1.6.
Asterisk ip 170.251.100.9 mizu ip 192.168.234.208
If i put a sofphone in 192.168.234.208 register with no problems, the network works fine. Now i put mizu in Apache server of 192.168.234.208, edit the Example.html and change: serveraddress: '170.251.100.9',
Open it on other computer (x-lite, ekiga and other sofphones works fine too), put the username and pasword, clic on connect and nothing happend, Register Failed. No logs in Asterisk also cli. If i put a destination and clic on call: == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 [May 23 10:15:38] WARNING[10285]: chan_sip.c:22682 stop_session_timer: Null stimer in stop_session_timer - 18e5811891207387329044k17518rmwp [May 23 10:15:38] WARNING[10285]: chan_sip.c:22682 stop_session_timer: Null stimer in stop_session_timer - 18e5811891207387329044k17518rmwp [May 23 10:15:38] WARNING[10285]: chan_sip.c:22682 stop_session_timer: Null stimer in stop_session_timer - 18e5811891207387329044k17518rmwp [May 23 10:15:38] WARNING[10285]: chan_sip.c:22682 stop_session_timer: Null stimer in stop_session_timer - 18e5811891207387329044k17518rmwp [May 23 10:15:38] WARNING[10285]: chan_sip.c:22682 stop_session_timer: Null stimer in stop_session_timer - 18e5811891207387329044k17518rmwp and nothing more
what i need to do to get work Mizu WEbphone with Asterisk?
If i use the chat button, i can see this on asterisk cli: [May 23 10:21:39] WARNING[10285]: chan_sip.c:14429 receive_message: Received message to <sip:9072@170.251.100.9> from <sip:800@170.251.100.9>;tag=44g8217721287307897459m, dropped it... Content-Type:text/plain Message: hello
Hello
Please try the last webphone build.
This should fix the problem (We have a lot of customers using the webphone with Astersik without any issue)
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