Brad Johns writes:
> Do you have "method_filtering" in registrar params set to 0 or 1?
i have
modparam("registrar", "method_filtering", 1)
and it worked ok with tryit client.
-- juha
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mai
Peter Dunkley writes:
> There is a much simpler WebSocket Kamailio configuration file in the
> examples directory in the source tree:
> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples/welcome.cfg;h=4bde0ae71be6f8da6a5bd3089d5af57569cc0178;hb=HEAD
the link to websocket
Sorry,
Wrong link. The correct one is:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples/websocket.cfg;h=4176af0a86985dc88d768b31f4ebe4021abb093f;hb=HEAD
Peter
> Hi,
>
> There is a much simpler WebSocket Kamailio configuration file in the
> examples directory in the so
Hi,
There is a much simpler WebSocket Kamailio configuration file in the
examples directory in the source tree:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples/welcome.cfg;h=4bde0ae71be6f8da6a5bd3089d5af57569cc0178;hb=HEAD
It doesn't have accounting or any of the other
Very interesting. I am still not able to have an audio call complete. Can
I see your kamailio.cfg, under separate cover? Or sent to the list?
Do you have "method_filtering" in registrar params set to 0 or 1? I had it
set to 1 and by default the JsSIP tryit must not have been sending an Allow
o
Peter,
Thank you. By changing the "method_filtering" modparam to 0 (it was
actually 1), I am now able to make it past this, and the INVITE is
processed over WS transport. However, the audio call is still not
completing.
I am seeing a "180 Ringing" message for a while, followed by a "408 Request
Juha Heinanen writes:
> Mar 29 10:04:13 sars /usr/sbin/sip-proxy[626]: : [dns_cache.c:3374]:
> BUG: sip_resolvehost: unknown proto 5
> Mar 29 10:04:13 sars /usr/sbin/sip-proxy[626]: ERROR: tm [ut.h:314]: ERROR:
> uri2dst: failed to resolve "tfq7u3ur8iqs.invalid" :bug - critical error (-13)
tha
when tryit app sends invite to my proxy, i get this to syslog:
Mar 29 10:04:13 sars /usr/sbin/sip-proxy[626]: INFO: INVITE
by as from <192.98.102.131> is authorized
Mar 29 10:04:13 sars /usr/sbin/sip-proxy[626]: INFO: registrar [lookup.c:313]:
instance is
Mar 29 10:04:13 sars /usr/sbin/sip-p
Juha Heinanen writes:
> by bad, i had not reconfigured WS URI and therefore the default proxy
> was used as outbound proxy.
>
> i'll try again.
now that tied with my own proxy where i had activated websocket
transport, tryit app worked fine. i was even able to hear voice and see
video.
does an
Juha Heinanen writes:
> i noticed that tryit app uses normal sip tcp port 5060. i didn't even
> have websocket.so loaded in my proxy when i did the test. perhaps it
> falls back to sip over tcp if sip over websocket transport is not
> supported?
by bad, i had not reconfigured WS URI and therefor
Brad Johns writes:
> However, after registration, the users can't place an audio call. I see no
> ringing on the remote browser. I don't know how to debug this further to
> find out what the problem is. Can anyone help with clues or debug? In
> Debug log I can see the websocket ws_frame.c deco
Hello,
In SIP you can put an Allow: header in REGISTER requests to say which
methods the registering end-point is capable of receiving.
If you get a -2 returned from lookup() it means that the method for the
request (in this case INVITE) was not in the "Allow:" header in the
REGISTER.
You can ch
Hi,
New to Kamailio. I have my Kamailio 4.0 server with websocket support, and
the users can register using the JsSIP Tryit sample WebRTC application.
They can do 'chat' feature of JsSIP Tryit using kamailio 4.0 sip over
websockets module.
However, after registration, the users can't place an a
Hi,
New to Kamailio. I have my Kamailio 4.0 server with websocket support, and
the users can register using the JsSIP Tryit sample WebRTC application.
However, after registration, the users can't place an audio call. I see no
ringing on the remote browser. I don't know how to debug this furthe
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