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

Alfred E. Heggestad alfred.heggestad at gmail.com
Sun Nov 27 18:13:52 CET 2016


On 11/27/16 12:53 PM, Juha Heinanen wrote:
> 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
> happens:
>
> 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: 0.0.0.0:10504, remote: 192.26.133.10:40018/192.26.133.10:40019
> 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]
>   seq_put=0
>

Hey Juha,

thanks for the report.

I made a fix for this, could you please try with this commit:

https://github.com/alfredh/baresip/commit/1203abd362a212b6cb6a02ff54f466d3a23dbf83


the menu module is still using dynamic menus; if there is one or more calls
a set of in-call related commands will be available. some people said that
they did not like this, other people said it was quite nice .. so I guess
it is up for debate :)




/alfred



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