Summary: | ASTERISK-08515: ALSA output causes Seg Fault | ||
Reporter: | edgreenberg (edgreenberg) | Labels: | |
Date Opened: | 2007-01-08 08:34:53.000-0600 | Date Closed: | 2007-02-15 20:03:35.000-0600 |
Priority: | Critical | Regression? | No |
Status: | Closed/Complete | Components: | Applications/app_dial |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | Was dialing from the console and call was connecting with OSS output. Switched modules.conf to noload=>oss (in order to use alsa) and got seg fault on audio output. ****** ADDITIONAL INFORMATION ****** Asterisk Ready. *CLI> dial 014089830588 The 'dial' command is deprecated and will be removed in a future release. Please use 'console dial' instead. *CLI> -- Executing [014089830588@local:1] Dial("ALSA/hw:0,0", "SIP/fr01/014089830588") in new stack -- Called fr01/014089830588 -- SIP/fr01-083e1508 is making progress passing it to ALSA/hw:0,0 [Jan 8 06:25:05] WARNING[21725]: chan_alsa.c:770 alsa_indicate: Don't know how to display condition 14 on ALSA/hw:0,0 -- SIP/fr01-083e1508 is ringing -- SIP/fr01-083e1508 answered ALSA/hw:0,0 << Console call has been answered >> [Jan 8 06:25:06] WARNING[21725]: utils.c:725 tvfix: warning too large timestamp -14221443.9699231 [Jan 8 06:25:06] WARNING[21725]: utils.c:725 tvfix: warning too large timestamp -14221443.9699231 Segmentation fault | ||
Comments: | By: edgreenberg (edgreenberg) 2007-01-08 08:36:10.000-0600 This is with Fedora Core 6 on Dell Latitude D810 laptop (dual core 2.33). By: Serge Vecher (serge-v) 2007-01-08 14:39:38.000-0600 ed, can you please upload a backtrace? By: edgreenberg (edgreenberg) 2007-01-08 16:15:56.000-0600 #0 __ast_read (chan=0x90928e8, dropaudio=0) at channel.c:2074 #1 0x08094e83 in waitstream_core (c=0x90928e8, breakon=0xc258b0 "", forward=0x811a23b "", rewind=0x811a23b "", skip_ms=0, audiofd=-1, cmdfd=-1, context=0x0) at file.c:1029 #2 0x080951ca in ast_waitstream (c=0x90928e8, breakon=0xc258b0 "") at file.c:1093 #3 0x00c252b4 in playback_exec (chan=0x90928e8, data=0xb7a27f28) at app_playback.c:434 #4 0x080be6c8 in pbx_extension_helper (c=0x90928e8, con=0x0, context=0x9092a68 "local", exten=0x9092ab8 "#", priority=1, label=0x0, callerid=0x0, action=E_SPAWN) at pbx.c:505 ASTERISK-1 0x080c036c in __ast_pbx_run (c=0x90928e8) at pbx.c:2245 ASTERISK-2 0x080c12ae in pbx_thread (data=0x90928e8) at pbx.c:2556 ASTERISK-3 0x080eb82b in dummy_start (data=0x9092d90) at utils.c:545 ASTERISK-4 0x433ee3db in start_thread () from /lib/libpthread.so.0 ASTERISK-5 0x4334806e in clone () from /lib/libc.so.6 By: Joshua C. Colp (jcolp) 2007-01-24 20:22:59.000-0600 The backtrace provided shows an issue in app_playback but I don't see it in use at all on the call you gave - can you please provide a full backtrace with a newer checkout? thread apply all bt - thanks. By: edgreenberg (edgreenberg) 2007-01-24 21:53:16.000-0600 I will try to get this for you by end of week. I have limited internet connectivity right now. By: edgreenberg (edgreenberg) 2007-02-09 12:52:55.000-0600 I have had trouble with this.. a few times, I 'updated' and couldn't get a working build. Now I have a working build. I will try to get a test for you by end of weekend. By: edgreenberg (edgreenberg) 2007-02-09 13:09:42.000-0600 OK, this seems to be resolved in the latest svn. There was a comment that stated that there was no reason to use app_playback, but this seems incorrect. When the call connects, the system plays a little chirp through app_playback. We learn -- every day. Pleaes close bug. By: Serge Vecher (serge-v) 2007-02-09 13:21:28.000-0600 can you please provide the details as to when is app_playback required (so this can be documented)? By: Joshua C. Colp (jcolp) 2007-02-15 20:03:35.000-0600 Since this is already fixed I'm closing out this bug. If you still have the info for serge feel free to add it, but I think you just misunderstood what I said... but that's okay! |