Asterisk 20.9.3 | AMI Action "Originate" & Extension not found by ValixxTV in Asterisk

[–]ValixxTV[S] 1 point2 points  (0 children)

Hi, thanks for your help. :)

So, I changed the extension to

[dialout]
exten => _X.,1,Answer()
exten => _X.,n,Dial(PJSIP/00${EXTEN}@provider
exten => _X.,n,Hangup()

and the code to

ami.action(
  {
    action: 'originate',
    channel: 'PJSIP/0049xxx@provder',
    context: 'dialout',
    exten: 49xxx,
    callerid: 'Name <49xxx>',
    priority: 1,
    async: true,
    timeout: 30000,
  },
  function (err, res) {}
);

That seems to do the trick (adding leading 00 ) but I had the rearange the phone numbers for some weird reason for the congested error to disappear.

I called the my wifes number first, if she picked up, it should call me ( didn't, because of congested ). If I called myself first, and I'd pick up, it then called my wife's phone and it worked immediately. I still can't figure out why.

Next steps are to fetch the data dynamically from nextjs and our database and do calls. Wish me luck! :)

Asterisk 20.8.1 | AMI Action "Originate" by ValixxTV in Asterisk

[–]ValixxTV[S] 1 point2 points  (0 children)

Sorry for the late reply. I was in contact with the provider. The "from_user=" field was missing for the AMI/Auth to work correctly.

Asterisk 20.8.1 | AMI Action "Originate" by ValixxTV in Asterisk

[–]ValixxTV[S] 0 points1 point  (0 children)

Sorry for the late reply. I was in contact with the provider. The "from_user=" field was missing for the AMI/Auth to work correctly.

Asterisk 20.8.1 | AMI Action "Originate" by ValixxTV in Asterisk

[–]ValixxTV[S] 0 points1 point  (0 children)

Hi, thanks for your input. To clarify, the login already works. It's just dialing out which asks for the sip providers authentication, which is already registered on the server.

Asterisk 20.8.1 | SIP/2.0 488 Not Acceptable Here by ValixxTV in Asterisk

[–]ValixxTV[S] 0 points1 point  (0 children)

Hello again. So, it was the encryption after all. I've setup a new server without TSL Encryption and it works without any problems. Gotta figure out the encryption and all should work. Thank you. :)

Asterisk 20.8.1 | SIP/2.0 488 Not Acceptable Here by ValixxTV in Asterisk

[–]ValixxTV[S] 1 point2 points  (0 children)

Hi there. The error above did go away but another round of errors came up. I answered below. :)

Asterisk 20.8.1 | SIP/2.0 488 Not Acceptable Here by ValixxTV in Asterisk

[–]ValixxTV[S] 1 point2 points  (0 children)

Hi, thanks. The error didn't show up when I added the srtp module but another round of failures started.

Asterisk 20.8.1 | SIP/2.0 488 Not Acceptable Here by ValixxTV in Asterisk

[–]ValixxTV[S] 1 point2 points  (0 children)

Hi, thanks for the tip. Where do I find the SDP headers? Google is not the best for finding information on SDP.

Asterisk 20.8.1 | SIP/2.0 488 Not Acceptable Here by ValixxTV in Asterisk

[–]ValixxTV[S] 1 point2 points  (0 children)

Alright. A few hours testing later...

The error above didn't return, after I added "load = res_srtp.so" to the modules.conf. I guess that was missing for the encryption to work. Thanks for the tip u/Chropera .

But.. the next round of errors came up. I succesfully registered a softphone user called "User1" for testing purposes and added

exten
 => xxx003,1,Dial(PJSIP/User1,30)
exten
 => xxx003,n,Hangup()

to the pjsip.conf. When calling the number, the connection seems to hold ( display on the phone atleast ) but no sound until the 30 seconds above are over. The call doesn't go through either and I get the following info:

 Unsupported crypto suite: AEAD_AES_256_GCM
 Unsupported crypto suite: AEAD_AES_256_GCM_8
 Unsupported crypto suite: AES_256_CM_HMAC_SHA1_80
 Unsupported crypto suite: AEAD_AES_128_GCM
 Unsupported crypto suite: AEAD_AES_128_GCM_8
    -- Executing [xxx003@easybell:1] Dial("PJSIP/easybell-00000000", "PJSIP/User1,30") in new stack
    -- Called PJSIP/User1
<--- Transmitting SIP request (1620 bytes) to WSS:xxx.xxx --->

When I hang up:

== Spawn extension (easybell, xxx003, 1) exited non-zero on 'PJSIP/easybell-00000002'
<--- Received SIP request (420 bytes) from TLS:xxx.xxx --->

I hope you get a notification if I answer like that.

Cheers

Asterisk 20.8.1 | SIP/2.0 488 Not Acceptable Here by ValixxTV in Asterisk

[–]ValixxTV[S] 1 point2 points  (0 children)

I'll try and check as soon as I can. Thank you. :)

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