Version 1.4.2 but to be honest I have no reason at all to suspect
that this is a problem with the asterisk software.
I've able to replicate this from a few different "client" net
connections and a across a few different linksys ata's. Where when
you call into the
host and enter the extension to connect to you miss the last digit of
the extension. Almost every time you miss the last digit of the
extension
(in a 4 digit extension). My suspicion is simply because of the
network we are currently using to host the asterisk box, as a packet
dump on the
lan segment clearly showed that the ATA transmitted all digits
(rfc2833) but the asterisk host only recieved 3 of the 4. The second
you dial
slower everything works fine; also the lines for "voice" are clear
with no noticeable impairments. I'm more curious if anyone else has
ever run
into a similar problem and what the resolution was if they found one
(IE a sturdier net connection for the asterisk host), or Tweaking
the timers
on the ata's to slow down how fast and how long they transmit
digits. I've done a few different tests and if I use a 'softphone'
dialing directly into
the server things work perfectly. I can dial as fast as I want,
however when I come in through the pstn trunks through the upstream
provider I find this problem.
has anyone else ever seen this? Or seen a case where mis-matched
dtmf modes across multiple providers causes this problem?
minor detail on what I referred to as the 'pstn trunks' I have no
analog or digital circuts all handoffs are sip.
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
Bryan Laird
Saving Lost Packets since 1994
Have you seen this packet? 1010101111010
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