asterisk users - May 2012 - No compatible codecs, not accepting this offer! - after upgrading to 1.8.11

Hi,

I've upgraded my asterisk 1.4 to the version 1.8.11. After making some
adjustments to the configuration files to port it to the new version, calls
between registered phones in asterisk, work fine, but inbound calls coming
from the SIP trunk I have with a telco to asterisk, don't work anymore. I
don't know why!...

This is the SDP portion that comes in the INVITE messages of calls through
that trunk (let's say, whose endpoint has the IP x.x.x.x, purposely
omitted). Nothing seems to be wrong with that to me:
v=0
o=CSM 0 1 IN IP4 x.x.x.x
s=Acme
c=IN IP4 x.x.x.x
t=0 0
m=audio 22152 RTP/AVP 8 0 18 4 101
a=rtpmap:101 telephone-event/8000

And here's the debugging:
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:5092 do_setnat: Setting NAT on RTP
to Off
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing
session-level SDP v=0... UNSUPPORTED.
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing
session-level SDP o=CSM 0 1 IN IP4 x.x.x.x... UNSUPPORTED.
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing
session-level SDP s=Acme... UNSUPPORTED.
[May 8 17:45:30] DEBUG[6444]: netsock2.c:134 ast_sockaddr_split_hostport:
Splitting 'x.x.x.x' into...
[May 8 17:45:30] DEBUG[6444]: netsock2.c:188 ast_sockaddr_split_hostport:
...host 'x.x.x.x' and port ''.
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing
session-level SDP c=IN IP4 x.x.x.x... OK.
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing
session-level SDP t=0 0... UNSUPPORTED.
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537
ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on
0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537
ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on
0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537
ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on
0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537
ast_rtp_codecs_payloads_set_m_type: Setting payload 4 based on m type on
0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537
ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on
0x416e25b0
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:9110 process_sdp: Processing
media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640
ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640
ast_rtp_codecs_payload_formats: Incorporating payload 4 on 0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640
ast_rtp_codecs_payload_formats: Incorporating payload 8 on 0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640
ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640
ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x416e25b0
[May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible
codecs, not accepting this offer!


Any help?

Thanks,
Ricardo.
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On Wednesday 09 May 2012, Ricardo Carvalho wrote:
> [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible
> codecs, not accepting this offer!
> 
> Any help?
Are you sure you compiled all the codecs you need?

What happens if you run `make menuselect` in both the 1.4 source tree and in 
the 1.8 source tree, "side-by-side" in tabs of the same terminal
window?  You
need at least GSM, A-law and micro-law.

(The above is my preferred method of building a configuration like an existing 
one.  No doubt someone will weigh in with a better way of doing it.)

-- 
AJS

Answers come *after* questions.