| Summary: | ASTERISK-23924: Don't know of any 0x0 (Nothing) formats | ||
| Reporter: | Ross Beer (rossbeer) | Labels: | |
| Date Opened: | 2014-06-24 11:32:37 | Date Closed: | 2014-07-14 08:43:21 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | 1.8.28.2 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | Cent OS 6 | Attachments: | |
| Description: | After upgrading to Asterisk 1.8.28.2 the CLI is showing lots of the following:
{noformat} [Jun 24 17:24:48] WARNING[50530]: channel.c:1115 ast_best_codec: Don't know any of 0x0 (nothing) formats [Jun 24 17:24:48] WARNING[50530]: channel.c:1115 ast_best_codec: Don't know any of 0x0 (nothing) formats {noformat} This issue is not happening on any other version, but is consistent over multiple boxes using the same realtime database. The boxes running 1.8.28.0 are not showing this message. I have noticed that the realtime structure has 'disallow=all' before 'allow=alaw;ulaw', would this cause an issue with any changes made in the latest release? | ||
| Comments: | By: Matt Jordan (mjordan) 2014-06-24 22:02:18.927-0500 I think your configuration is invalid - codecs should be delineated with a {{,}}, not a {{;}}. You may have a WARNING message when first loading that indicates that. By: Ross Beer (rossbeer) 2014-06-25 16:16:12.409-0500 Changing ';' to ',' has not resolved the issue. I have discovered that the issue occurs when a calls is made as follows: PSTN (g722,alaw,ulaw,gsm) --> Asterisk 1 (g722,alaw,ulaw,gsm,h263,h263p,h264) --> Asterisk 2 (g722,alaw,ulaw,gsm,h263,h263p,h264) --> Phone (alaw,ulaw,gsm,h263,h263p,h264) The calls passing from the PSTN is 'g722' and the phone only accepts 'alaw', therefore I would expect the call to be re-invited with the new codec. However Asterisk 2 is transcoding g722 <-> alaw The call is setup and two way audio is sucessful. The issue doesn't occur if a call is made from the phone to the PSTN directly via Asterisk 2. Asterisk 1 does play a prompt before passing the call to Asterisk 2. By: Matt Jordan (mjordan) 2014-06-25 16:45:50.226-0500 We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information Go ahead and make sure 'sip set debug on' is enabled as well. A dump of the SIP peers from your database would help as well. By: Rusty Newton (rnewton) 2014-07-14 08:43:13.209-0500 Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines | ||