we have a asterisk-based calling card platform, using three different sip originations (such as IPKall service, that someone dials a DID and the pstn call is delivered in sip/rtp format to our server).
two of the three sip origis are working fine, but one is not due to poor dtmf delivery. i’m using the word poor because ~80 digits out of 100 pressed are delivered while 20 are lost.
we’ve tried different codec setting between 729 and 711. but no avail.
i’m wondering what settings and configs should be looked into on our side as well as theirs? the sip orig provider is using StarNet.