Config Calling for WEB ( HTTP )

I need to make a call on the web,
I already config the archive manager.conf, but every time i go to call, this error show up
=> WARNING [20182]: channel.c: 6137 request_channel: No channel type registered for ‘SIP’
and realy this channel dont resgistered in archive sip, why i am using PJSIP
My question is, how i go calling on the web with PJSIP ??

There is a guide on the wiki for configuring Asterisk for WebRTC[1] using PJSIP. You will still need a client in the browser.


It appears you are specifying the channel type for chan_sip, not that for chan_pjsip.

If you are initiation the call using the Asterisk manager, it is not a Webrtc call. I assume you have specified the wrong channel on the originate action

I’m following the guide tab on the wiki, but it’s not working
my configuration in http.conf >>>

enabled = yes
bindaddr =
bindport = 8088
tlsenable = yes
tlsbindaddr = 8089
tlscertfile = /etc/asterisk/keys/asterisk.crt,/etc/asterisk/keys/asterisk.csr,/etc/asterisk/keys/asterisk.pem
tlsprivatekey = /etc/asterisk/keys/asterisk.key

and my output to the commands in the CLI >>>

HTTP server status:
Server: Asterisk / 16.1.1
Server enabled and linked to

URIs enabled:
/ httpstatus => General Asterisk HTTP Status
/ static / … => Asterisk HTTP static delivery

Allowed redirects:

does not look the same on the wiki and I do not know why
can you help me?

I dont see the link for the guide you have followed

Asterisk manager interface it is not used when using webrtc, I used webrtc on the past I stopped using it, for me it is not a reliable solution as things keep changing constantly and many technologies involved. a lot work to make things works and sunddenly they stop working. Of course this is my personal opinion

okay, but how to use the file manager using the PJSip command if the configurator asks for SIP information, as it is using the asterisk with mysql (real-time)?

Sorry I don’t understand your question

how do I configure Manager.conf using PJSIP?
because I’m using real-time asterisk

This question does not make sense.

I’ll explain better

my Manager.conf >>
enabled = yes
;webenabled = yes
port = 5038
bindaddr =

secret = teste
permit =
read = system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan
write = system,call,agent,user,config,command,reporting,originate,message

every time I try to call I get the error -> WARNING [20182]: channel.c: 6137 request_channel: No channel type registered for ‘SIP’


the error is understandable since my channels are being registered in a mysql database
in Pjsip format using odbc
ps_endpoints => odbc, asterisk
ps_aors => odbc, asterisk
and so on.
the question is to be able to make connections via the web, without changing the configuration already made to dial between the channels
Can you understand?

If you want to initiate a call using a web client to bridge 2 channels, you can do it using AMI and Originate Action, but that it is nothing related to the Webrtc guide you posted, if what you want is using the Web Broswer as a SIP client then you have to use webrtc

And if you re using AMI let me see how do you initiate the call, let me see the originate command syntax

I’m not using,
How do I use AMI?
Can you help me with the setup?

manager.conf is totally irrelevant to the error you are getting.

I got it, it’s making calls through the web, but I still have a problem that I do not know how to solve
Even though you are making the call between the extensions, the connection is muted, do you know if a socket is missing? I’m plugin and how should it be done?