RTP Timeout Question

Asterisk Asterisk SIP
1

Hello,

We’re running Asterisk 20.15.0 and are experiencing an issue where a school administrator is trying to record an update to their message line, which is hosted by a third party. The message line number is 888-100-1000.

The administrator, at extension 100, places the call. The call goes out through the trunk to the PSTN, and they follow the prompts to enter their PIN and begin recording. However, as soon as the recording starts, the third party stops sending RTP.

(Call flow: PJSIP/100 → Asterisk → PJSIP/8881001000@Upstream → PSTN)

Initially, I had the RTP timeout on the trunk endpoint set to 30 seconds, which I later increased to 180 seconds. However, since the endpoint for extension 100 still has a 30-second timeout, the call drops after that time.

Is there a way to configure Asterisk to ignore RTP timeouts when calling certain numbers? Or that it should honor the 180?

Please let me know if you have any questions.

Thanks,

2

I rather suspect it is the phone that you need to tell to ignore RTP timeouts. As far as I know, RTP timeouts only apply to received, not to sent, RTP.

There might be an option to send silent media, to the phone.

3

I stand corrected. It times out because of the settings on the Trunk, it does honor and times out after 180s.

Asterisk does support sending silence during recording, but it works the other way around.