Webrtc outbound call works, incoming call at webrtc client does not work

rebell21 · September 14, 2025, 7:16pm

I have created a webrtc client for openhab. the sip client in openhab shall call my doorbird using SIP and the doorbird shall call openhab. This requires a websocket communication.

I can call my doorbird from the webrtc client . But when the doorbird (or for test purposes my zoiper from the notebook) calls the endpoint webrtc_client (openhab), it doesn’t work

I get the error message Everyone is busy/congested at this time (1:0/1/0).

The asterisk-cli output

    -- Executing [6005@internal:1] NoOp("PJSIP/notebook-00000039", "Call openhab") in new stack
    -- Executing [6005@internal:2] Dial("PJSIP/notebook-00000039", "PJSIP/webrtc_client") in new stack
    -- Called PJSIP/webrtc_client
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [6005@internal:3] NoOp("PJSIP/notebook-00000039", "Status: 34") in new stack
    -- Executing [6005@internal:4] Hangup("PJSIP/notebook-00000039", "") in new stack
  == Spawn extension (internal, 6005, 4) exited non-zero on 'PJSIP/notebook-00000039'
<--- Transmitting SIP response (395 bytes) to UDP:192.168.2.185:60531 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.2.185:60531;rport=60531;received=192.168.2.185;branch=z9hG4bK-524287-1---5bb6e9a07e8c3ba2
Call-ID: CZDEXsd3bib4CoH5mZ8JXg..
From: "notebook" <sip:notebook@speedport.ip>;tag=41c8a479
To: <sip:6005@speedport.ip>;tag=b4cc24dc-3c4f-4bf5-8cca-6a5750429162
CSeq: 2 INVITE
Server: AsteriskTZ
Reason: Q.850;cause=34
Content-Length:  0

I call the sipclient of openhab for tests from the notebook using the extension 6005.

here is my extensions.conf

exten => 6001,1,NoOp(Call Doorbird)
same =>  n,Dial(PJSIP/doorbird,20)
same =>  n,NoOp(Status: ${HANGUPCAUSE})
same =>  n,Hangup()

exten => 6002,1,NoOp(calling desktop)
same =>  n,Dial(PJSIP/zoiper,20)
same =>  n,Hangup()

exten => 6003,1,NoOp(calling notebook)
same =>  n,Dial(PJSIP/zoiper,20)
same =>  n,Hangup()

exten => 6004,1,NoOp(Group call from Doorbird)
same => n,Dial(PJSIP/zoiper&PJSIP/notebook&PJSIP/webrtc_client,20)
same => n,Hangup()

exten => 6005,1,NoOp(Call openhab)
same => n,Dial(PJSIP/webrtc_client)
same => n,NoOp(Status: ${HANGUPCAUSE})
same => n,Hangup()

here is my pjsip,conf with the webrtc endpoint

[authentication](!)
type=auth             ; type of section: authentication
auth_type=userpass    ; password authentication

[aor_template](!)
type=aor              ; find out where the endpoint can be contacted
max_contacts=1

; Definitions of user accounts associated with equipment

[doorbird](endpoint_basic)
auth=doorbird
aors=doorbird
callerid="Doorbird" <6001> ; to have the name of the caller displayed
[doorbird](authentication)
password=doorbird
username=doorbird
[doorbird](aor_template)

[notebook](endpoint_basic)
auth=notebook
aors=notebook
callerid="Notebook" <6003>
[notebook](authentication)
password=notebook
username=notebook
[notebook](aor_template)

[webrtc_client]
type = identify
match = 192.168.2.0
endpoint = webrtc_client

[webrtc_client]
type=aor
max_contacts=5
remove_existing=yes
contact=sip:webrtc_client@dynamic

[webrtc_client]
type=auth
auth_type=userpass
username=webrtc_client
password=webrtc_client

[webrtc_client]
type=endpoint
aors=webrtc_client
auth=webrtc_client
dtls_auto_generate_cert=yes
webrtc=yes
outbound_auth=webrtc_client
;transport=transport-wss
context=internal
disallow=all
allow=ulaw,opus

asterisk-PBX*CLI> pjsip show contacts

  Contact:  <Aor/ContactUri..............................> <Hash....> <Status> <RTT(ms)..>
==========================================================================================

  Contact:  doorbird/sip:doorbird@192.168.2.131:5060;ob    f5f5dd59f7 NonQual         nan
  Contact:  notebook/sip:notebook@192.168.2.185:60531;tran 3ac263029f NonQual         nan
  Contact:  speedport/sip:192.168.2.1:5060                 45977b638f NonQual         nan
  Contact:  webrtc_client/sip:webrtc_client@dynamic        c8cb8ced14 NonQual         nan

Objects found: 4

why is there no channel available ?

collinks79 · September 14, 2025, 7:46pm

I have this problem too. I usually restart asterisk. This is just a temporary solution. I am hopeful that an expert will give us a permanent solution

COLLINS ONYEGBADO | B.Tech; Msc | CCNA; CCNP
Head, Hardware Maintenance Operations & Networking Unit.
Mobile: +2348064550911 | Voip:4001 | www.fuotuoke.edu.ng
ICT CENTER
FEDERAL UNIVERSITY OTUOKE, BYELSA STATE

On Sun, Sep 14, 2025, 20:27 rebell21 via Asterisk Community <notifications@asterisk.discoursemail.com> wrote:

	rebell21 September 14

I have created a webrtc client for openhab. the sip client in openhab shall call my doorbird using SIP and the doorbird shall call openhab. This requires a websocket communication.

I can call my doorbird from the webrtc client . But when the doorbird (or for test purposes my zoiper from the notebook) calls the endpoint webrtc_client (openhab), it doesn’t work

I get the error message Everyone is busy/congested at this time (1:0/1/0).

The asterisk-cli output

    -- Executing [6005@internal:1] NoOp("PJSIP/notebook-00000039", "Call openhab") in new stack
    -- Executing [6005@internal:2] Dial("PJSIP/notebook-00000039", "PJSIP/webrtc_client") in new stack
    -- Called PJSIP/webrtc_client
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [6005@internal:3] NoOp("PJSIP/notebook-00000039", "Status: 34") in new stack
    -- Executing [6005@internal:4] Hangup("PJSIP/notebook-00000039", "") in new stack
  == Spawn extension (internal, 6005, 4) exited non-zero on 'PJSIP/notebook-00000039'
<--- Transmitting SIP response (395 bytes) to UDP:[192.168.2.185:60531](http://192.168.2.185:60531) --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.2.185:60531;rport=60531;received=192.168.2.185;branch=z9hG4bK-524287-1---5bb6e9a07e8c3ba2
Call-ID: CZDEXsd3bib4CoH5mZ8JXg..
From: "notebook" <sip:notebook@speedport.ip>;tag=41c8a479
To: <sip:6005@speedport.ip>;tag=b4cc24dc-3c4f-4bf5-8cca-6a5750429162
CSeq: 2 INVITE
Server: AsteriskTZ
Reason: Q.850;cause=34
Content-Length:  0

I call the sipclient of openhab for tests from the notebook using the extension 6005.

here is my extensions.conf

exten => 6001,1,NoOp(Call Doorbird)
same =>  n,Dial(PJSIP/doorbird,20)
same =>  n,NoOp(Status: ${HANGUPCAUSE})
same =>  n,Hangup()

exten => 6002,1,NoOp(calling desktop)
same =>  n,Dial(PJSIP/zoiper,20)
same =>  n,Hangup()

exten => 6003,1,NoOp(calling notebook)
same =>  n,Dial(PJSIP/zoiper,20)
same =>  n,Hangup()

exten => 6004,1,NoOp(Group call from Doorbird)
same => n,Dial(PJSIP/zoiper&PJSIP/notebook&PJSIP/webrtc_client,20)
same => n,Hangup()

exten => 6005,1,NoOp(Call openhab)
same => n,Dial(PJSIP/webrtc_client)
same => n,NoOp(Status: ${HANGUPCAUSE})
same => n,Hangup()

here is my pjsip,conf with the webrtc endpoint

[authentication](!)
type=auth             ; type of section: authentication
auth_type=userpass    ; password authentication

[aor_template](!)
type=aor              ; find out where the endpoint can be contacted
max_contacts=1

; Definitions of user accounts associated with equipment

[doorbird](endpoint_basic)
auth=doorbird
aors=doorbird
callerid="Doorbird" <6001> ; to have the name of the caller displayed
[doorbird](authentication)
password=doorbird
username=doorbird
[doorbird](aor_template)

[notebook](endpoint_basic)
auth=notebook
aors=notebook
callerid="Notebook" <6003>
[notebook](authentication)
password=notebook
username=notebook
[notebook](aor_template)

[webrtc_client]
type = identify
match = 192.168.2.0
endpoint = webrtc_client

[webrtc_client]
type=aor
max_contacts=5
remove_existing=yes
contact=sip:webrtc_client@dynamic

[webrtc_client]
type=auth
auth_type=userpass
username=webrtc_client
password=webrtc_client

[webrtc_client]
type=endpoint
aors=webrtc_client
auth=webrtc_client
dtls_auto_generate_cert=yes
webrtc=yes
outbound_auth=webrtc_client
;transport=transport-wss
context=internal
disallow=all
allow=ulaw,opus

why is there no channel available ?

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rebell21 · September 14, 2025, 8:05pm

restarting asterisk doesn’t solve the problem in my case

david551 · September 14, 2025, 10:57pm

My guess would be that the webrtc client has dropped its TLS connection. I don’t think that a webrtc server can start a connection, in the same way as an HTTP/HTML web server can’t.

Having a 1 in the second element of the list, means that one of the called destinations (you only have the one) was congested, and 34 means no circuits available. Combined with the lack of any resulting SIP logs, it seems to mean that there was no way to connect. (This paragraph is accurate. The first paragraph is my current understanding, but I’ve no actual experience with WebRTC.)

rebell21 · September 15, 2025, 10:39am

after I have installed ctxSip on my client and called from my zoiper softphone by extension 6005, it suddenly worked again.

I have installed ctxSip as webrtc client and called by the browser with the URL https://192.168.2.182:8443/static/phone (openhab web server). In my zoiper on the notebook I dialed extension 6005. I could call ctxSip as webrtc client from zoiper as extension 6005. This call was routed by the asterisk websocket server. Then I killed the page with the ctxSip webrtc client and called from zoiper again the extension 6005. Now openhab signaled the incoming call and I could accept the call. The connection was established and I could hear the audio. Strange. Did the ctxSip re-establish a dropped connection between the client and the asterisk server ? Did the ctxSip phone fix something on the client side?

system · October 15, 2025, 10:40am

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.

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Webrtc outbound call works, incoming call at webrtc client does not work

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