Jitter Buffer - SIP to E1 PSTN

[dj_ndc]

Hello

I am using Yate with Sangoma as SIP to PSTN Gateway, how to veryfy if jitterbuffer is enabled in configuration ?
I have problem with fax transmission when source transmission came to Yate without jitterbuffering, so I need to enable/determine jitter buffer on Yate, I have set:

yrtpchan.conf

tos=cs0
buffer=200
minjitter=120
maxjitter=200

Greetings

[dj_ndc]

Hello

I have figured out, it was my mistake. I have used old regexroute plan for Cisco SLT with rtp forwarding to Cisco Gateways, so I had:

${rtp_forward}possible=;rtp_forward=yes

I have changed it to rtp_forward=no, and now it works fine, and live changes in yrtpchan.conf works after reload.

Greetings