Sipml5.org Asterisk : SIP 200 Response when Register, but SIP 401 and 403 when Dial

[Din Assegaf]

Hi, I am using sipml5.org for testint web-> telephony

currently here .. http://doubango.org/sipml5/call.htm?svn=179

Disabled Video

and Enabled RTCWeb Breaker

I see 'SIP/2.0 200' onto UDP socket destined for 188.165.231.30:13060

When SIP register, its ok

but Dial I see

'SIP/2.0 401' onto UDP socket destined for 188.165.231.30:13060

Failed to authenticate device <sip:r...@XX.XX.XX.XX>;tag=393562300

[May  1 05:12:06] DEBUG[7932]: chan_sip.c:3684 __sip_xmit: Trying to put 'SIP/2.0 403' onto UDP socket destined for 188.165.231.30:13060

[May  1 05:12:14] DEBUG[7932]: chan_sip.c:26041 handle_request_do: Invalid SIP message - rejected , no callid, len 448

I test with X-Lite no problem, so any hint what to do ?

-

Syeh Abidin

[navaismo]

Seems like You are using Asterisk, so the 401 is a normal response when you are trying to do a call, then the phone need to respond with the ACK and  another INVITE with the result of the challenge in order to allow the call.

About the other message:

[May  1 05:12:14] DEBUG[7932]: chan_sip.c:26041 handle_request_do: Invalid SIP message - rejected , no callid, len 448

Seems like Asterisk(or maybe a route issue) lost or destroyed the callid  and cant send the other  SIP messages.

 

[Din Assegaf]

On Wed, May 1, 2013 at 11:21 PM, navaismo <nava...@gmail.com> wrote:

Seems like You are using Asterisk, so the 401 is a normal response when you are trying to do a call, then the phone need to respond with the ACK and  another INVITE with the result of the challenge in order to allow the call.

About the other message:

[May  1 05:12:14] DEBUG[7932]: chan_sip.c:26041 handle_request_do: Invalid SIP message - rejected , no callid, len 448

Seems like Asterisk(or maybe a route issue) lost or destroyed the callid  and cant send the other  SIP messages.

 

I see, but still at the end of sipml5 just got 403 Forbidden Response.

No idea, looks like I will try find another scenario like sipml5->web2rtc->sipproxy (new, opensips) -> asterisk media server ...

until soon asterisk fully support webrtc ..

 

[navaismo]

Asterisk support WebRTC just need to be configured you have the sipml5 wiki and asterisk wiki too, to do that. You can also paste the full sip debug to see how is the call progress.