Condition Warning

[alexanderzhirov]

Linux: Alpine 3.16.3
Asterisk: 18.13.0
3G Modem Huawei E1550
chan_dongle channel driver for Huawei UMTS cards

I assembled this Asterisk + old GSM modem assembly. Everything works! SMS messages are sent and received, USSD works, incoming and outgoing calls also work. Everything is just chic and sparkle!

Since I’m only new to working with Asterisk, I have a question. I sometimes get a warning like this:

	...
    -- Started music on hold, class 'default', on channel 'Dongle/dongle0-0100000000'
[Nov 26 22:01:50] WARNING[2167][C-00000002]: channel.c:1079 channel_indicate: [Dongle/dongle0-0100000000] Don't know how to indicate condition 22
    -- PJSIP/1991-00000000 is ringing
    ...

I think it makes no sense to throw the whole log. There are no errors and other warnings before this warning.

In this regard, I have a question: what does this warning mean and what does condition 22 mean? Where can I see information about the conditions in general and find out what is going wrong?

[david551]

chan_dongle is third party code, and I believe it is abandoned by its developer.

The message means that chan_dongle does not implement support for connected line updates, which seems very likely as mobile phones don’t change their number mid-call.

The key to these conditions can be found at:

asterisk/asterisk/blob/master/include/asterisk/frame.h#L277

[alexanderzhirov]

@david551

chan_dongle is third party code, and I believe it is abandoned by its developer.

Yes, the developer most likely abandoned this code. But I use a fork that works quite well with the latest versions of Asterisk.

Anyway, thank you for the answer!

Is there a way to skip this condition check? Or is it already being solved at the level of the chan_dongle module itself?

[david551]

You should ignore the warning, although it seems to be misrepresenting the file and function name involved. (It is probably coming from chan_dongle itself, as similar messages for other technologies come from the respective channel drivers.)

The Asterisk core will pass the indication to the channel driver in the same way that it passes all indications, and it is up to the driver to decide whether it can be acted upon. A SIP driver would generate a re-INVITE, or UPDATE with a revised P-Asserted-Identity or Remote-Party-ID-Header. As the mobile phone air interface doesn’t expect anything except a phone, it isn’t going to provide a method of doing anything similar.

[alexanderzhirov]

Thanks!