Summary: | ASTERISK-18239: crash in 1.8 | ||
Reporter: | Private Name (falves11) | Labels: | |
Date Opened: | 2011-08-08 10:53:54 | Date Closed: | 2011-08-11 16:56:29 |
Priority: | Critical | Regression? | |
Status: | Closed/Complete | Components: | Channels/chan_sip/General |
Versions: | 1.8.5.0 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | Red Hat 64 Bits latest update | Attachments: | ( 0) 18crash.txt ( 1) 18crash2.txt |
Description: | I just ported my app from 1.6.2 and in 1.8, it crashes nonstop. | ||
Comments: | By: Private Name (falves11) 2011-08-08 11:51:50.621-0500 keeps crashing By: Leif Madsen (lmadsen) 2011-08-08 12:47:11.447-0500 You say you just ported your application from 1.6.2 to 1.8.... what application? By: Private Name (falves11) 2011-08-08 12:57:03.220-0500 When I say "I just ported my application", I mean my dialplan. Not a real Asterisk application, just a particular dialplan that worked fine in 1.6.2. I will be happy to upload it if you let me know how to do it so it remains confidential. By: Paul Belanger (pabelanger) 2011-08-08 13:56:51.107-0500 If you are able to reproduce this, upload an example dialplan to the issue. You have been told multiple times when reporting problems you must include detailed information about it. Stating, 'crash in 1.8' and 'it crashes nonstop' are no longer acceptable and will result in your issue just being blindly closed. If you want the community to help fix your problems, then please provided them the information they need. Having us keep asking you every time is not productive. By: Private Name (falves11) 2011-08-08 14:10:38.715-0500 exten =>_X.,n,Dial(${CUT(CARRIERLIST,-,${i})},${TO},aM(get-callid^${PROMPT})L(10800000)) However, I detected a very bad additional problem with 1.8. There is no way to force it do local bridging, versus "remote bridging". It behaves different than 1.6.2. No caller from behind a NAT can get audio both ways. The only way is to make them send a different codec so Aterisk is forced to transcode. If the codec is the same for both legs of the call, Asterisk ALWAYS chooses to remove itself from the media, and this any NAT callers are killed. I tried everything in the book. By: Jason Parker (jparker) 2011-08-11 15:11:44.495-0500 Is this actually 1.8.5.0? What patches have you applied? What modifications have you made? By: Private Name (falves11) 2011-08-11 15:25:37.176-0500 This the command that I use svn co http://svn.digium.com/svn/asterisk/branches/1.8 asterisk No patches or any modifications. By: Jason Parker (jparker) 2011-08-11 16:56:29.723-0500 Revision 331578 should fix your problem. |