I installed the webrtcphone on Freepbx.
When i make a call between 2 user and i got an error :
[2015-09-21 11:11:36] WARNING[22784][C-00000010]: app_dial.c:2421 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/99401
– Connected line update to SIP/99402-0000001b prevented.
– SIP/99401-0000001c is ringing
[2015-09-21 11:11:40] ERROR[21385][C-00000010]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo(“nvboprff2gof.invalid”, “(null)”, …): Name or service not known
[2015-09-21 11:11:40] WARNING[21385][C-00000010]: chan_sip.c:16053 __set_address_from_contact: Invalid host name in Contact: (can’t resolve in DNS) : ‘nvboprff2gof.invalid’
i tried to change some value but it still had the same error !
I used Asterisk (Ver. 12.8.2) with pjproject and installed the Webrtcphone on freepbx 12.
it’s ok when i make a call between 2 softphone.
The peer is not Asterisk. This cannot be fixed from the Asterisk end.
The peer is currently sending the domain name nvboprff2gof.invalid. It needs to be reconfigured so that it sends a domain name or IP address that will actually reach it.
FreePBX uses SIPJS to create the webrtc sip client and according the SIPJS documentation you nned a setting to allow calls in asterisk. Chek if that parameter is set.
thanks for your answer.
sorry about my stupid but how do i set the sip.js ?
i read this guide and i know it is my problem but i dont know where sip.js file is .
From sipjs.com/guides/server-configuration/asterisk/ :
"Configure SIP.js
Asterisk does not accept Contact headers with the .invalid domain. When creating a UA, add the configuration parameter hackIpInContact. If you are missing this property you will be able to make calls from WebRTC, but not receive calls through Asterisk will fail.
Additionally this guide will only work with audio calls, Asterisk will reject video calls.
The following configuration example creates a UA for the Asterisk configuration above. Replace the values with the values from your config.
var config = {
// Replace this IP address with your Asterisk IP address
uri: ‘1060@127.0.0.1’,
// Replace this IP address with your Asterisk IP address,
// and replace the port with your Asterisk port from the http.conf file
ws_servers: ‘ws://127.0.0.1:8088/ws’,
// Replace this with the username from your sip.conf file
authorizationUser: ‘1060’,
// Replace this with the password from your sip.conf file
password: ‘password’,
// HackIpInContact for Asterisk
hackIpInContact: true,
// Invite with audio only
ua.invite(‘1061’,{
media: {
constraints: {
audio: true,
video: false
}
}
});
i see the jssip.js file in the path : /var/www/html/admin/modules/webrtc/ucp/assets/jssiplibs/
and the guide on the link you give me said that i need to change from jssip libary to sip.js libary, can you show me how to do that. i googled but dont have a answer.
i’m newbie so i sorry about my question and my english.
Thanks for your help !
i have install a new system. asterisk 13 and freepbx 12.
the call was successful when i made a call between 1 softphone and 1 webrtcphone.
But between 2 webrtcphone wasn’t.
The old error :
[2015-09-24 20:32:48] ERROR[10045][C-00000014]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo(“hsdpphrqj3ef.invalid”, “(null)”, …): Name or service not known
[2015-09-24 20:32:48] WARNING[10045][C-00000014]: chan_sip.c:16151 __set_address_from_contact: Invalid host name in Contact: (can’t resolve in DNS) : ‘hsdpphrqj3ef.invalid’