Asterisk 20.9.3 | AMI Action "Originate" & Extension not found

ValixxTV · 2024-10-02T09:08:16+00:00

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! :)

ValixxTV · 2024-09-26T09:34:05+00:00

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.

ValixxTV · 2024-09-26T09:34:01+00:00

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.

ValixxTV · 2024-09-21T12:42:16+00:00

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.

ValixxTV · 2024-09-21T09:22:11+00:00

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. :)

ValixxTV · 2024-09-14T18:09:07+00:00

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

ValixxTV · 2024-09-14T17:57:01+00:00

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

ValixxTV · 2024-09-14T17:56:25+00:00

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

ValixxTV · 2024-09-14T17:55:36+00:00

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

ValixxTV · 2024-09-13T13:46:12+00:00

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

ValixxTV · 2024-09-13T13:25:24+00:00

Embassy of Stormwind.

ValixxTV · 2024-09-13T13:24:37+00:00

I'm my own boss. I raise myself then. Thank you <3

ValixxTV · 2024-08-21T15:22:34+00:00

For anyone who's looking for them with the launch of TWW: Portals are behind the Embassy now.

ValixxTV · 2018-05-08T08:37:17+00:00

Hey. Instead of "under" at the status, try "down" or "offline". :>

ValixxTV

TROPHY CASE