[Asterisk-Users] Outgoing Calls

[Michael]

I am trying to route my calls through an outside IAX provider.  I am having
a problem with which codec to use.  The only way I have successfully been
able to make an outgoing call is if i do:

  disallow=all
  allow=g729

in the sip.conf file (for my phones) and the iax.conf file.  The second I
add one more codec to that list, for instance:

  disallow=all
  allow=g729
  allow=ulaw

I get the following error in the CLI:

  Nov 23 10:56:35 NOTICE[3799]: channel.c:1703 ast_set_write_format: Unable
to find a path from ulaw to g729
  Nov 23 10:56:35 NOTICE[3799]: channel.c:1736 ast_set_read_format: Unable
to find a path from g729 to ulaw

During this time, the number I am calling rings, however, when I pick up,
the server hangs up and says this:

  -- IAX2/plainvoip/3 is ringing
  -- IAX2/plainvoip/3 stopped sounds
  -- IAX2/plainvoip/3 answered SIP/4035-0e93
  Nov 23 10:56:41 WARNING[3799]: channel.c:2127 ast_channel_make_compatible:
No path to translate from SIP/4035-0e93(4) to IAX2/plainvoip/3(256)
  Nov 23 10:56:41 WARNING[3799]: app_dial.c:1024 dial_exec: Had to drop call
because I couldn't make SIP/4035-0e93 compatible with IAX2/plainvoip/3
  -- Hungup 'IAX2/plainvoip/3'
  == Spawn extension (from-sip, 13102801234, 1) exited non-zero on
'sip/4035-0e93'

 I will post my configuration files if necessary.  Thank you in advance for
any help that anyone can offer.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051123/372e47dd/attachment.htm