Summary:ASTERISK-08542: G729 No path to translate.
Reporter:David Faulk (dfaulk)Labels:
Date Opened:2007-01-10 09:55:03.000-0600Date Closed:2011-06-07 14:08:25
Versions:Frequency of
Description:We have two Asterisk servers, one running 1.2 and one running 1.4 both connected to the same SIP VOIP provider. On the 1.2 server, until the call is answered  we can see 0/0 encoders and  decoders in use , that switches to 1/1 once th call is answered. During a call on the 1.4 server we see 1/1 encoders and decoders and while the call is ringing then we get this error:
~~~~ Call is ringing ~~~~
This is the 1.4 server.

CLI> show g729

1/1 encoders/decoders of 1 licensed channels are currently in use

[Jan 9 11:57:19] WARNING[25731]: rtp.c:874 ast_rtcp_read: RTCP Read too

[Jan 9 11:57:24] WARNING[25731]: rtp.c:874 ast_rtcp_read: RTCP Read too

~~~~ Call is answered ~~~~

-- SIP/icallglobe-081f74f8 answered SIP/lukas-b774dc90

[Jan 9 11:57:24] WARNING[25731]: channel.c:3033
ast_channel_make_compatible: No path to translate from
SIP/lukas-b774dc90(524290) to SIP/icallglobe-081f74f8(256)

[Jan 9 11:57:24] WARNING[25731]: app_dial.c:1592 dial_exec_full: Had to
drop call because I couldn't make SIP/lukas-b774dc90 compatible with

Comments:By: Brandon Kruse (bkruse) 2007-01-10 15:38:05.000-0600


I think this is related to the stuff we have been working on, right?

By: Jason Parker (jparker) 2007-01-16 15:32:16.000-0600

bkruse, file: What is the status of "that other bug"?

dfaulk3: Are you still seeing this with the latest svn branch 1.4?

By: Brandon Kruse (bkruse) 2007-01-16 19:05:51.000-0600

Qwell: I am not 100% sure on what exactly file is trying to do.

But the problem has been logged internally and is getting worked on

can you send a reminder to mog to tell him to take a look at this??

I think it is the same problem we are seeing


By: lukaso (lukaso) 2007-01-16 19:47:51.000-0600

I'm the person who reported the bug. I don't have the ability to test on the latest SVN (because the vendor I was connecting to has disabled the connection). Also, is the codec available on SVN? Or does latest refer to the rest of asterisk 1.4?

By: Joshua C. Colp (jcolp) 2007-02-15 20:21:22.000-0600

lukaso: The codec is not available from SVN, but Asterisk 1.4 is. That is what was meant by checking the latest SVN. If you would still like to try to track this down though and have access to the resources then the total console output would be helpful too. If not then we can close this.

By: Russell Bryant (russell) 2007-02-21 17:43:39.000-0600

What type of SIP phone is being used here?

By: Russell Bryant (russell) 2007-02-21 19:02:22.000-0600

I fixed something that may resolve this issue.  Please re-test and see if you still have a problem.

The fix was in rev 56011 in the 1.4 branch.

By: lukaso (lukaso) 2007-02-21 21:12:12.000-0600

I am unable to test the issue as I no longer have access to an account with g729 codec support.

The SIP client was wengophone which was using GSM/H323 codecs.

By: Russell Bryant (russell) 2007-02-22 10:19:08.000-0600

Ok, well thanks for the update.  I'm going to suspend this issue and consider it fixed unless someone can reproduce it.