Grandstream HT-841 and Asterisk 20.2.1

Asterisk Asterisk SIP
1

Hi
This gateway has 4 FXO and 1 FXS port.

I can make inbound and outbound calls on the FXS port and inbound calls from PSTN to the FXO port. When I make outbound calls from an extension to the FXO port outbound it fails (see attached SIP dialog, PJSIP, EXTENSIONS files).

I used the ‘peer’ model for this media gateway and do not use SIP Registration.

I suspect that I have the Grandstream HT-841 setup incorrectly, but would like to know if the Asterisk end makes sense.

I have AudioCodes media gateways working for many years with SIP, not PJSIP. The switch to PJSIP and Grandstream media gateways taken a long time to learn. I am resisting the urge to go back to SIP…

I would publish a simple guide to the forum when this gets working, if that is of interest.
Thanks
K Melden
Failed FXO call.txt (6.4 KB)
GS support extensions.txt (805 Bytes)
GS support PJSIP.txt (446 Bytes)

2

Based on your SIP logs, the device is replying with a final SIP answer ‘SIP/2.0 486 Busy Here,’ and it is very explicit why it is returning that answer: ‘Warning: 399 GS All channels are in use.’ Therefore, I suggest you start working with the settings and support forum, because Asterisk is correctly sending the INVITE request. There is nothing you can do except hang up active calls to release the channels in use in the SIP gateway, but I don’t think this is the case here.

3

Fixed. HT-841 default ports for FXO are 6060, 6062, etc. Helps to read the configuration carefully.
k

4

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