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?
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Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
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