Asterisk 13.13.1 WebRTC Issue

Hello guys so i’m doing a sipml5 integration with Asterisk 13.13.1 and it requires a secure websocket connection. I can connect the phones to the Asterisk Server and I can call from them but I can’t accept calls. The issues is that something apparently sets the address of the device to .invalid. I dug into the issue and found this forum link but the problem is that when I go to /var/www/html/admin/modules/webrtc/ucp/assets/jssiplibs I have no sip.js file. Anyone able to help? Thanks in advance. Any help will be greatly appreciated.

You are using SIPML5 and the post is for JSSIP users. Better share the log from browser and Asterisk with the settings.

Logging in phone 1 from browser :
Websocket connection from ‘’ for protocol ‘sip’ accepted using version '13’
Registered SIP ‘6008’ at
Saved useragent “IM-client/OMA1.0 sipML5-v1.2015.03.18” for peer 6008

-Identical output for SIP’6009’

Calling ‘6009’ from ‘6008’ from browser :
DTLS ECDH initialized(secp256r1), faster PFS enabled
Using SIP RTP CoS mark 5
Executing[6009@outgoing:1] Dial(“SIP/6008-000000”,“SIP/6009”) in new stack
DTLS ECDH initialized(secp256r1), faster PFS enabled
Using SIP RTP CoS mark 5
[Dec 23 11:35:15] ERROR[2513][C-0000000000]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo(“df7jal23ls0d.invalid”, “(null)”, …) Name or service not known
[Dec 23 11:35:15] WARNING[2513][C-000000000]: chan_sip.c:16610 __set_address_from_contact: Invalid host name in Contact: (can’t resolve in DNS) :'df7jal23ls0d.invalid’
Called SIP/6009
SIP/6009-0000001 is ringing
Got SIP response 603 “Failed to get local SDP” back from
SIP/6009-000000000001 is busy
Everyone is busy/congested at this time (1:1/0/0)
Auto fallthrough, channel ‘SIP/6008-00000000’ status is busy

I should probably mention that I installed AsteriskNOW on the virtual machine which apparently comes with FreePBX GUI

Show us the settings of the peer please.