[Re-devel] cannot answer call with latest baresip/re

Juha Heinanen jh at tutpro.com
Sun Nov 27 12:53:20 CET 2016

i first built and installed latest libre.  then i built and installed
latest baresip.  when i try to answer call (which used to be earlier
possible by pressing enter and now seem require pressing 'a' key), this

sip:jh at test.tutpro.com: Incoming call from: Test sip:test at test.tutpro.com - (press 'a' to accept)

--- Audio stream ---
 tx:     ptime=20ms
 rx:     ptime=20ms pt=-1
 audio tx pipeline:         src ---> encoder
 audio rx pipeline:        play <--- decoder
 audio dir=sendrecv pt_enc=-1
 local:, remote:
RTP debug:
 Encode: seq=9437 ssrc=0xe876d104
----- RTCP Session: -----
  cname= SSRC=0xe876d104/3900100868 rx=0Hz
  TX: packets=0, octets=0
--- jitter buffer debug---
 running=0 min=5 cur=0 max=10 [frames]

-- juha

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