Summary:	ASTERISK-30093: res_pjsip_refer: Pickup event is not sent from refer_incoming_invite_request
Reporter:	Jasper Hafkenscheid (jasper.hafkenscheid)	Labels:	patch
Date Opened:	2022-06-02 06:46:01	Date Closed:
Priority:	Minor	Regression?
Status:	Open/New	Components:	Resources/res_pjsip_refer
Versions:	18.11.2	Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) debug.log ( 1) full-cel.csv ( 2) res_pjsip_refer_call_pickup_stasis.patch
Description:	Similar to ASTERISK-28081, it seems that the stasis/cel PICKUP event is not sent when a call-pickup is performed using the INVITE with Replaces method. I have tried splicing in a function call to send_call_pickup_stasis_message, this fixes our tests that rely on this event. I suspect a better approach is possible by using the ast_do_pickup method, but have not gotten that to work.
Comments:	By: Asterisk Team (asteriskteam) 2022-06-02 06:46:02.471-0500 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: Kevin Harwell (kharwell) 2022-06-02 09:58:00.779-0500 Please provide relevant dialplan and configuration for replicating the problem. As well as steps to reproduce. An Asterisk log with SIP tracing/debug {{pjsip set logger on}} enabled could be helpful too. Thanks! By: Jasper Hafkenscheid (jasper.hafkenscheid) 2022-06-03 07:42:27.533-0500 I setup asterisk 18.12.1 with the default basic-pbx configuration. Scenario: - 1101 registers - 1102 sends an invite to 1101 - 1101 replies 180 and forwards details to 1103 - 1103 sends an invite with replaces - 1101 receives a cancel (without the expected 'answered elsewhere') - 1103 and 1102 get bridged - call gets ended as expected CEL logs do not show the expected PICKUP event. By: Jasper Hafkenscheid (jasper.hafkenscheid) 2022-06-16 01:30:53.681-0500 In the provided patch I exposed the `send_call_pickup_stasis_message` method from `pickup.c` and spliced it into `refer_incoming_invite_request` from `res_pjsip_refer.c`. This feels wrong, duplicating the code and exposing its internals. I think a better way would be to hook into `ast_do_pickup` from `pickup.c`, but have not gotten that to work. By: Jasper Hafkenscheid (jasper.hafkenscheid) 2022-07-19 04:14:19.752-0500 Are there any updates on this ticket? I notice it is linked to a Digium internal ticket, but I can't read that. By: Joshua C. Colp (jcolp) 2022-07-19 04:16:10.636-0500 Any updates would already be posted on this issue. It's unassigned, thus noone is likely working on it, but is open. There are also no other updates or comments on the Sangoma internal ticket.