Summary:	ASTERISK-30499: inband_progress not honored with ivr/direct accept
Reporter:	Jørgen H (jorgen)	Labels:
Date Opened:	2023-04-20 03:15:01	Date Closed:	2023-04-20 06:36:49
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_pjsip
Versions:	20.2.0	Frequency of Occurrence	Constant
Related Issues:
Environment:	Linux x64	Attachments:
Description:	According to 20.2 documentation, setting inband_progress=no on the other partys endpoint should prevent Asterisk from sending "183 Session Progress" to it. It is however sent, and at least if Asterisk picks up the call directly using IVR/audio file playback Editing line 1678 in channels/chan_pjsip.c changing case AST_CONTROL_PROGRESS by adding if (channel->session->endpoint->inband_progress && ast_channel_state(ast) != AST_STATE_UP)  response_code = 183; solves the problem, but I'm not sure if this breaks something else. The reason for us not sending 183 is that there is a new bug in the SIP implementation of a commercial provider causing race condition and hangup triggered by 183.
Comments:	By: Asterisk Team (asteriskteam) 2023-04-20 03:15:05.578-0500 WARNING JIRA will be going read-only at the end of April, 2023. We will be starting fresh on Github at https://github.com/asterisk/asterisk at that time. No issues or patches will be copied to Github. If you file an issue on JIRA at this time you will need to recreate it on Github after this date. The same applies if you have a patch. WARNING 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: Joshua C. Colp (jcolp) 2023-04-20 04:37:31.305-0500 That's not what the option is actually for, and changing it as such would be a substantial breaking change. The option controls whether ringing is sent as inband progress. It's not to stop all session progress - such a thing would have to be behind a new option - because that can fundamentally just break calls and give them no audio. By: Joshua C. Colp (jcolp) 2023-04-20 04:38:29.577-0500 Will you be creating an option to implement such behavior? If so, I'd suggest waiting until we move to Github. By: Jørgen H (jorgen) 2023-04-20 06:35:16.863-0500 I see. I'm not sure about the need for an additional option just for this behaviour since it is due to a third party regression bug. Im not familiar with the need for such an option in generally. I expect the third party/commercial provider to deploy a bugfix, but due to the size of this company it will probably take a while If they have not fixed it within some time I can continue with this issue at Github.