Summary:	ASTERISK-29233: faxdetect_timeout does not work with channel SIP
Reporter:	Duc (ducto)	Labels:	fax
Date Opened:	2021-01-05 01:31:12.000-0600	Date Closed:	2021-01-05 03:55:14.000-0600
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	Channels/chan_dahdi Channels/chan_sip/General
Versions:	13.30.0	Frequency of Occurrence	Constant
Related Issues:
Environment:		Attachments:
Description:	In version 13, in ChangeLog there is info about faxdetect_timeout option 2016-07-18 16:16 +0000 [652130feb2]  Richard Mudgett <rmudgett@digium.com>        * chan_dahdi: Add faxdetect_timeout option.          The new option allows the channel driver's faxdetect option to timeout on          a call after the specified number of seconds into a call.  The new feature          is disabled if the timeout is set to zero.  The option is disabled by          default. but as tested, it does not work properly using channel SIP. I tried to do the search and seemly it only support PJSIP root@dev-server:~# grep -rl faxdetect_timeout /usr/src/asterisk-13.30.0 /usr/src/asterisk-13.30.0/channels/chan_dahdi.h /usr/src/asterisk-13.30.0/channels/chan_dahdi.c /usr/src/asterisk-13.30.0/channels/misdn_config.c /usr/src/asterisk-13.30.0/channels/chan_dahdi.o /usr/src/asterisk-13.30.0/channels/chan_misdn.c /usr/src/asterisk-13.30.0/channels/chan_pjsip.c /usr/src/asterisk-13.30.0/channels/chan_dahdi.so /usr/src/asterisk-13.30.0/channels/dahdi/bridge_native_dahdi.o /usr/src/asterisk-13.30.0/include/asterisk/res_pjsip.h /usr/src/asterisk-13.30.0/include/asterisk/res_fax.h /usr/src/asterisk-13.30.0/ChangeLog /usr/src/asterisk-13.30.0/res/res_fax.c /usr/src/asterisk-13.30.0/res/res_fax.o /usr/src/asterisk-13.30.0/res/res_fax_spandsp.so /usr/src/asterisk-13.30.0/res/res_fax_spandsp.o /usr/src/asterisk-13.30.0/res/res_pjsip/pjsip_configuration.c /usr/src/asterisk-13.30.0/res/res_fax.so /usr/src/asterisk-13.30.0/configs/samples/chan_dahdi.conf.sample /usr/src/asterisk-13.30.0/CHANGES Can you please kindly check? Thank you, Duc
Comments:	By: Asterisk Team (asteriskteam) 2021-01-05 01:31:14.194-0600 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: Asterisk Team (asteriskteam) 2021-01-05 01:31:14.698-0600 Per the Asterisk versions page [1], the maintenance (bug fix) support for the Asterisk branch you are using has ended. For continued maintenance support please move to a supported branch of Asterisk. After testing with a supported branch, if you find this problem has not been resolved, please open a new issue against the latest version of that Asterisk branch. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions By: Duc (ducto) 2021-01-05 01:34:12.087-0600 Even thought the bug report is for my testing Asterisk version 13, I tried with latest Asterisk version 18.1.1 and still have the same issue: [root@dev-server]# grep -rl faxdetect_timeout asterisk-18.1.1/ asterisk-18.1.1/include/asterisk/res_fax.h asterisk-18.1.1/include/asterisk/res_pjsip.h asterisk-18.1.1/channels/chan_misdn.c asterisk-18.1.1/channels/chan_dahdi.c asterisk-18.1.1/channels/chan_pjsip.c asterisk-18.1.1/channels/chan_dahdi.h asterisk-18.1.1/channels/misdn_config.c asterisk-18.1.1/configs/samples/chan_dahdi.conf.sample asterisk-18.1.1/CHANGES asterisk-18.1.1/res/res_fax.c asterisk-18.1.1/res/res_pjsip/pjsip_configuration.c asterisk-18.1.1/ChangeLog By: Asterisk Team (asteriskteam) 2021-01-05 01:34:12.289-0600 This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable. By: Joshua C. Colp (jcolp) 2021-01-05 03:55:15.114-0600 The chan_sip module does not have such functionality implemented. It would have to be added as a new feature. The chan_sip module is also community supported and receives little attention, as well we do not currently allow feature requests on the issue tracker.