Didn’t change anything for node-js on the systems.
This is incorrect. The webrtc phone rings simultaneously with its “parent” extension (the extension without the 99).
@billsimon As long as it’s in User/Device mode for Chan_SIP. It treats it as a second device which is used by the user. It’s very clear in the Wiki documentation for WebRTC that this is what happens. I’ve also tested it.
You can register as many endpoints as you want via Chan_SIP but if you’re just in Extensions mode it will send the calls to the last endpoint that send a valid REGISTER. You need to be in User/Device to send calls to multiple devices with the same extension.
I have tested it also and it works as I have said. Please link to the documentation you are talking about.
There is no registration overlap. My main extension registers as YYY and my webrtc phone registers as 99YYY. FreePBX dialplan has logic to call both and I don’t have to set device-and-user mode.
Are you using Chan_SIP or Chan_PJSIP?
http://wiki.freepbx.org/display/FPG/WebRTC+Phone-UCP
Overview
The WebRTC Module allows an Administrator to enable a “WebRTC phone” that can be attached to a user’s extension which they can connect to through FreePBX User Control Panel, this WebRTC phone will then receive phone calls at the same time as the users extension using user and device mode behind the scenes. If you have User and Device Mode enabled any extension you enable the WebRTC Phone a duplicate extension of 99XXXX will be created (where XXXX is the original extension number), when the user then views the web interface of the WebRTC phone they will be connected to device 99XXXX which will receive calls from the original extension
Chan_sip. That documentation is outdated - February 2015. A lot has changed since then, and now it is pretty much seamless to add a webrtc extension. You don’t need to be in User and Device mode. If you are in regular Extension mode it does add the 99xxx device and uses dialplan to simultaneously ring both.
Then I will double check that.
In my case, when I dial webRTC phone from other extension, I receive “Service Unavailable”.
with and without 99.
I check the webRTC extension settings too :
- Name : 99302
Description :
Secret :
MD5Secret :
Remote Secret:
Context : from-internal
Record On feature : automon
Record Off feature : automon
Subscr.Cont. :
Language : en
Tonezone :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Named Callgr :
Nam. Pickupgr:
MOH Suggest :
Mailbox : 99302@device
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : “302” <99302>
MaxCallBR : 384 kbps
Expire : 151
Insecure : no
Force rport : No
Symmetric RTP: No
ACL : Yes
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Path support : No
Path : N/A
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 192.168.10.100:49650
Defaddr->IP : (null)
Prim.Transp. : WS
Allowed.Trsp : WS,WSS
Def. Username: 9pn2bn2s
SIP Options : outbound
Codecs : (ulaw|alaw|gsm|g726)
Auto-Framing : No
Status : UNREACHABLE
Useragent : SIP.js/0.7.5
Reg. Contact : sip:[email protected];transport=wss
Qualify Freq : 60000 ms
Keepalive : 0 ms
Sess-Timers : Refuse
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : Yes
Are you sure?
This is why it does not ring.
This is probably your mismatch.
Dear I didn’t do any manually configuration. You did?
No. I am using HTTPS. Chrome and Firefox both work for me.
My mistake about the mismatch. Mine shows Prim.Trans.: WS also but I am connected over WSS.
You are not reading this correct. It is stating if you are in extension mode your webrtc is setup as another device of your primary extension and you can not call devices direct. You can only call users which is your primary extension hence if your extension is 4002 and you enable WebRTC your WebRTC device is 994002 but you call 4002 and both will ring.
How this works ! I’m confused.
any special configuration?
after login to UCP, I see in asterisk :
[2016-06-19 08:10:40] NOTICE[26301]: chan_sip.c:29921 sip_poke_noanswer: Peer ‘99302’ is now UNREACHABLE! Last qualify: 0
what is your asterisk version? mine is 13.9.1
Any idea ?
How can I debug node UCP ?
Checkup this: http://wiki.freepbx.org/display/SUP/Providing+Great+Debug
Maybe it helps
Only one error I see in Asterisk logs:
This is related to WSS
[2016-06-19 16:14:20] ERROR[21084] tcptls.c: SSL_shutdown() failed: 1
Asterisk 13.9.1 has related bug?
As I requested before, turn on sip debugging for this extension and post logs of the registration process.
I did an update on WebRTC phone and certificate manager to edge version. now via http, web phone is registered but when I make a call I receive this:
WARNING[6543][C-00000000]: chan_sip.c:10709 process_sdp: Can’t provide secure audio requested in SDP offer
So this update turned off my road to the goal and I don’t have any WebRTC on http/https.
I found a bad config too, when I login to UCP via http and webRTC phone is registered, Asterisk sends Notify message via WSS, and because that he doesn’t receive any answer, makes my webRTC extension “Unreachable”.
I remove WSS from webRTC phone settings and it now my phone is reachable.
but other issues still exist .
May I report last bad setting in issue tracker.
I did a downgrade and with removing “WSS” from extension settings, now I can call and called but only via HTTP and using Firefox.
Please try this:
remove WSS from your webRTC extension settings:
/etc/asterisk/sip_additional.conf
Hi psdk,
as Billsimon mentioned, can you debug the extension and post here, I also would be interested, as I had similar issues and downgraded my WebRTC … which cant be the final Solution.
I cant upgrade at the Moment, as I have users using it, but if you can, maybe upgrade again and debug.
sip set debug on/off
Log files you will find in: /var/log/asterisk
thanks, NUB