Asterisk Codec negotiation problem

Asterisk Asterisk SIP
1

Hi, i’m trying to make some calls between SIP 181 and 182

Some relevant environment info:
Asterisk v15.2.2
181 supports G711, G729 codecs
182 supports only G729 codec

The configuration is the same for both:
sip.conf

context=phones
type=friend
host=dynamic
disallow=all
allow=alaw
allow=ulaw
allow=g729

But when SIP 182 answers the call, asterisk fails in this errors:

  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
    -- Executing [181@phones:1] NoOp("SIP/182-00000010", "Iniciando ligação para 181") in new stack
    -- Executing [181@phones:2] Dial("SIP/182-00000010", "SIP/181") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/181
    -- SIP/181-00000011 is ringing
[May 29 10:43:23] WARNING[1817][C-0000000a]: channel.c:5522 set_format: Unable to find a codec translation path: (alaw) -> (g729)
[May 29 10:43:23] WARNING[1817][C-0000000a]: channel.c:5522 set_format: Unable to find a codec translation path: (g729) -> (alaw)
    -- SIP/181-00000011 answered SIP/182-00000010
[May 29 10:43:23] WARNING[3971][C-0000000a]: channel.c:6480 ast_channel_make_compatible_helper: No path to translate from SIP/181-00000011 to SIP/182-00000010
[May 29 10:43:23] WARNING[3971][C-0000000a]: app_dial.c:3176 dial_exec_full: Had to drop call because I couldn't make SIP/182-00000010 compatible with SIP/181-00000011
  == Spawn extension (phones, 181, 2) exited non-zero on 'SIP/182-00000010'

Can someone help me or explain what i’m doing wrong?
Thanks in advance.

2

Asterisk has no backwards path for codecs selections.

3

I think your asterisk box doesn’t have a valid g729 codec.
please run : core show translation recalc

and see if there any valid codec translation path between 711 and 729

4

I think he is expecting the restriction to G.729 to feed back to the input channel. He probably knows he is in a G.729 pass through configuration but expected Asterisk to look at the global picture and limit both sides to G.729, but Asterisk doesn’t look at the outgoing side when setting up the incoming one.