Carlos Chavez wrote:
I had an old Asterisk installation die recently and we decided to
upgrade to Asterisk 13 to replace the old server. Everything seems to be
working with PJSIP but there is one issue. Asterisk talks to a
callmanager via a SIP trunk and send calls to PSTN (another country).
Most of the calls that go through there are for conferences. Desk phones
can enter the conferences without any issues but users with softphones
like Zoiper cannot. The conference systems either duplicate digits or
drop some. I have tried using inband, info and rfc4733 but the
softphones always have the same problem.
Anyone has any experience with softphone dtmf issues?
While I can't say I've had problems I can say you might want to try to
narrow things down a bit. Do testing strictly to Asterisk first using
something like Read and see if that is fine. If so then you've narrowed
it down to the outgoing side, which may mean that the length of digits
or something else produced by the softphone is not liked by it.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
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