Asterisk has no codecs?

Asterisk Asterisk Support
1

I’m debugging a larger issue, but right now I can’t figure out why my asterisk thinks there are no codecs available even though it’s already processing SIP calls on other trunks.

I’m setting up an IAX webphone and I get the following error:
[quote]NOTICE[22735]: chan_iax2.c:10967 socket_process: Rejected connect attempt from x.x.x.x,
requested/capability ‘0x2 (gsm)’/‘0x60e (gsm|ulaw|alaw|speex|ilbc)’ incompatible with our capability ‘0x0 (nothing)’.[/quote]

I have the usual disallow=all then allow=gsm and/or allow=ulaw and alaw, but I get the same error. It still says “our capability nothing”.

I’ve googled this to death and the only solution is to review and change the disallow and allow statements. I’ve tried just having allow=all and the same error.

Any help would be greatly appreciated.

2

What does:

core show translation recalc 60

from the Asterisk CLI give you?

3

[code]*CLI> core show translation recalc 60
Recalculating Codec Translation (number of sample seconds: 60)

     Translation times between formats (in microseconds) for one second of data
      Source Format (Rows) Destination Format (Columns)

       g723   gsm  ulaw  alaw g726aal2 adpcm  slin lpc10  g729 speex  ilbc  g726  g722 siren7 siren14 slin16  g719 speex16 testlaw
 g723     -     -     -     -        -     -     -     -     -     -     -     -     -      -       -      -     -       -       -
  gsm     -     -   465   465     1948   665   449  2215     -     -     -  1948   998      -       -   1997     -       -     482
 ulaw     -  1249     -    16     1532   249    33  1799     -     -     -  1532   582      -       -   1581     -       -      66
 alaw     -  1249    33     -     1532   249    33  1799     -     -     -  1532   582      -       -   1581     -       -      66

g726aal2 - 1865 665 665 - 865 649 2415 - - - 2148 1198 - - 2197 - - 682
adpcm - 1299 99 99 1582 - 83 1849 - - - 1582 632 - - 1631 - - 116
slin - 1216 16 16 1499 216 - 1766 - - - 1499 549 - - 1548 - - 33
lpc10 - 2199 999 999 2482 1199 983 - - - - 2482 1532 - - 2531 - - 1016
g729 - - - - - - - - - - - - - - - - - - -
speex - - - - - - - - - - - - - - - - - - -
ilbc - - - - - - - - - - - - - - - - - - -
g726 - 1865 665 665 2148 865 649 2415 - - - - 1198 - - 2197 - - 682
g722 - 2182 982 982 2465 1182 966 2732 - - - 2465 - - - 999 - - 999
siren7 - - - - - - - - - - - - - - - - - - -
siren14 - - - - - - - - - - - - - - - - - - -
slin16 - 3265 2065 2065 3548 2265 2049 3815 - - - 3548 1083 - - - - - 2082
g719 - - - - - - - - - - - - - - - - - - -
speex16 - - - - - - - - - - - - - - - - - - -
testlaw - 1232 32 32 1515 232 16 1782 - - - 1515 565 - - 1564 - - -[/code]

4

Yeah…your Asterisk install does seem to have GSM translation capability. Is the call coming in on an unmatched peer so the general section settings are being applied, not the peer-specific?

5

I have tried adding “allow=all” to both the general section and the user specific section (and made sure any disallow statements are removed). Problem persists…

6

That’s really bizarre. Dumb question time: you’ve done a CLI reload since making those change in the configuration file, right?

7

yes…and started/stopped asterisk…and rebooted…

8

bump…problem persists

9

This thread was an extension of this thread:
http://forums.digium.com/viewtopic.php?f=13&t=79610&start=0&hilit=IAX2+authentication+error&sid=2e975b3ef5116589ddeb9e28685effb7
…which is now resolved.

It appears the error reporting no codecs wasn’t an accurate response from asterisk on the root cause of my problems. The main thread has more details, but this thread is now a non issue.

Thank you for everyone’s input and assistance.