| Summary: | ASTERISK-15786: Data Buffer Size Exceeded! | ||
| Reporter: | Keisha Wynn (castleb) | Labels: | |
| Date Opened: | 2010-03-10 15:30:10.000-0600 | Date Closed: | 2011-06-07 14:00:46 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | PBX/pbx_config |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | All of our phone calls have been fading in and out almost like the person on the other end is going through a tunnel. It sounds like a cell phone call being disrupted. This is causing a lot of issues within our company with leads not hearing us talk and we are having to repeat ourselves numerous times. Other error messages are showing within our system that have not shown before. I have attached the error log below. Need help ASAP! ****** ADDITIONAL INFORMATION ****** 50 03/10/2010 4:08 PM WARNING chan_sip.c: This function can only be used on SIP channels. 51 03/10/2010 4:08 PM WARNING Last Message Repeated 3 times 52 03/10/2010 4:08 PM WARNING res_agi.c: Running DeadAGI on a live channel will cause problems, please use AGI 53 03/10/2010 4:07 PM WARNING chan_sip.c: This function can only be used on SIP channels. 54 03/10/2010 4:07 PM WARNING Last Message Repeated 3 times 55 03/10/2010 4:07 PM WARNING res_agi.c: Running DeadAGI on a live channel will cause problems, please use AGI 56 03/10/2010 4:07 PM WARNING chan_sip.c: This function can only be used on SIP channels. 57 03/10/2010 4:07 PM WARNING Last Message Repeated 3 times 58 03/10/2010 4:06 PM WARNING res_agi.c: Running DeadAGI on a live channel will cause problems, please use AGI 59 03/10/2010 4:06 PM WARNING Last Message Repeated 1 times 60 03/10/2010 4:05 PM WARNING ast_expr2.fl: If you have questions, please refer to doc/channelvariables.txt in the asterisk source. 61 03/10/2010 4:05 PM WARNING Last Message Repeated 2 times 62 03/10/2010 4:05 PM WARNING ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected $end, expecting '-' or '!' or '(' or '<token>'; Input: 63 03/10/2010 4:05 PM WARNING ast_expr2.fl: If you have questions, please refer to doc/channelvariables.txt in the asterisk source. 64 03/10/2010 4:05 PM WARNING Last Message Repeated 2 times 65 03/10/2010 4:05 PM WARNING ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected $end, expecting '-' or '!' or '(' or '<token>'; Input: 66 03/10/2010 4:05 PM ERROR pbx.c: Data Buffer Size Exceeded! 67 03/10/2010 4:05 PM WARNING res_agi.c: Running DeadAGI on a live channel will cause problems, please use AGI 68 03/10/2010 4:05 PM WARNING Last Message Repeated 1 times app_queue.c: The device state of this queue member, Local/7167@queue/o, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. | ||
| Comments: | By: snuffy (snuffy) 2010-03-10 16:27:42.000-0600 To help with this we would require a look at your dialplan as u seem to be getting errors from it. without the dialplan section in question and understanding the flow of the call we are unable to help.Thank you for the bug report however this report lacks information we need to investigate it further. For this reason, we are now going to close the bug - please feel free to reopen when you have more information at hand. By: Keisha Wynn (castleb) 2010-04-01 14:13:27 not able to look at dialplan. we do not have accessibility to the switchvox server. I am willing to give you as much information I can in order to get this resolved. By: Keisha Wynn (castleb) 2010-04-01 14:18:42 our system administrator has talked to Hao about this issue. I do know that Hao has logged into our system to check out some bug issues. I have also sent some sound files to support@switchvox.com Hao Duong Digium, Inc. | Switchvox Technical Support 7920 Arjons Drive, Unit E. San Diego, CA. 92126 - USA Toll Free: 1 888-881-3560 Support: +1 858-457-8299 Fax: +1 858-225-0271 support@switchvox.com By: Keisha Wynn (castleb) 2010-04-22 08:24:54 https://issues.asterisk.org/view.php?id=10209 By: Keisha Wynn (castleb) 2010-04-22 08:25:49 here is something that I have discovered, I am going to need this patch, but it is something you guys have to do for us since we do not have control over our server. I need this done ASAP, please let me know what I need to do By: Leif Madsen (lmadsen) 2010-04-22 11:16:34 Since this is a Switchvox server you need to contact Digium for support with this issue. The open source Asterisk tracker is for issues related to stock Asterisk compiled from source. I'm going to close this issue because this is not the appropriate venue for Switchvox support. Thanks! | ||