Asterisk 1.4.29 tl.74 MTE
all my customer have polycom phones (aside from a few ata's for faxes) but 2, 1 runs on aastra phones and 1 with linksys ata and both have dtmf issues (maybe they are the only one complaining about it) actually on the ata it is almost 0% success,
I do see the dtmf in dtmf.log and I have done a trace where dtmf goes trou, my carrier is claiming the reason is due to a 200 ms delay from the rtp packet to the dtmf packet and that is when the dtmf gets lost, they are suggesting to change to SipInfo.
1. Anyone have experience with SipInfo
2. What could the reason of delay (only on aastra and spa 2102)
Any help is appreciated
i think its related to a bug,
i think its related to a bug, but you're running 1.4.29 specifically to stop some crashes. see one of my other threads regarding dtmf and digium bugs... one of those bugs talks about it getting merged even into the 1.4 branch.
sip info is not working either so im trying to figure out how to reduce the delay
any thoughts?