Summary:	ASTERISK-29246: chan_ooh323, dtmf transit problem
Reporter:	Dmitry Melekhov (slesru)	Labels:
Date Opened:	2021-01-15 00:24:43.000-0600	Date Closed:
Priority:	Minor	Regression?
Status:	Open/New	Components:	Addons/chan_ooh323
Versions:	18.1.0	Frequency of Occurrence	Constant
Related Issues:
Environment:	Oracle Linux 8	Attachments:	( 0) dump.dump.gz ( 1) dump1.dump.gz
Description:	Hello! I migrated our centos6 and asterisk 13 installation to oracle linux 8 and asterisk installation several days ago using the same configuration as before. Yesterday user complained that he can't reach conference room on another side's cisco cucm and to enter this room he have to enter dtmf code, which is not recognized. This is how our systems are connected: PBX<--isdn pri-->asterisk<--h323-->ast-neftisa<---h323-->cucm I got traffic dump on ast-neftisa and I see no dtmf code passed to cucm address 10.77.6.21, I'll upload traffic dump. But if I set asterisk call flow to sip between asterisk, then I see dtmf in dump and everything works as expected PBX<--isdn pri-->asterisk<--sip-->ast-neftisa<---h323-->cucm Unfortunately, I have no idea what is changed between asterisk 13 and 18, but looks like bug in chan_oo323 for me..
Comments:	By: Asterisk Team (asteriskteam) 2021-01-15 00:24:44.039-0600 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur. Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/]. By: Dmitry Melekhov (slesru) 2021-01-15 00:27:17.434-0600 traffic when asterisks are connected using h323 By: Dmitry Melekhov (slesru) 2021-01-15 00:27:52.527-0600 traffic when asterisk are attached using sip By: Dmitry Melekhov (slesru) 2021-01-15 00:46:31.985-0600 btw, if I use SIP I see dtmf in debug: [Jan 15 10:44:57] DTMF[34849][C-00000017]: channel.c:3980 __ast_read: DTMF begin '5' received on SIP/asterisk-00000004 [Jan 15 10:44:57] DTMF[34849][C-00000017]: channel.c:3991 __ast_read: DTMF begin passthrough '5' on SIP/asterisk-00000004 [Jan 15 10:44:58] DTMF[34849][C-00000017]: channel.c:3866 __ast_read: DTMF end '5' received on SIP/asterisk-00000004, duration 180 ms [Jan 15 10:44:58] DTMF[34849][C-00000017]: channel.c:3917 __ast_read: DTMF end accepted with begin '5' on SIP/asterisk-00000004 [Jan 15 10:44:58] DTMF[34849][C-00000017]: channel.c:3955 __ast_read: DTMF end passthrough '5' on SIP/asterisk-00000004 [Jan 15 10:44:58] DTMF[34849][C-00000017]: channel.c:3980 __ast_read: DTMF begin '1' received on SIP/asterisk-00000004 [Jan 15 10:44:58] DTMF[34849][C-00000017]: channel.c:3991 __ast_read: DTMF begin passthrough '1' on SIP/asterisk-00000004 [Jan 15 10:44:58] DTMF[34849][C-00000017]: channel.c:3866 __ast_read: DTMF end '1' received on SIP/asterisk-00000004, duration 240 ms [Jan 15 10:44:58] DTMF[34849][C-00000017]: channel.c:3917 __ast_read: DTMF end accepted with begin '1' on SIP/asterisk-00000004 [Jan 15 10:44:58] DTMF[34849][C-00000017]: channel.c:3955 __ast_read: DTMF end passthrough '1' on SIP/asterisk-00000004 if I use ooh323- no :-( By: Dmitry Melekhov (slesru) 2021-01-15 03:05:41.237-0600 OK, more info - inband or rfc2833 does not work in case h323-h323 transit, h245signal or h245alphanumeric works, but only if it is set globally, setting let's say dtmfmode=inband in peer configuration changes nothing... :-( So, good- now I have working workaround, bad- something is wrong with inband ans rfc2833 methods. Thank you! By: Joshua C. Colp (jcolp) 2021-01-15 04:27:07.466-0600 [~may213] Would you like to triage this issue?