Summary: | ASTERISK-30129: Use pre-dial and post-dial (bridge) handlers together in Dial command only executes the pre-dial handler | ||
Reporter: | Fabian Borot (fborot) | Labels: | |
Date Opened: | 2022-07-06 14:13:53 | Date Closed: | 2022-07-06 15:05:41 |
Priority: | Critical | Regression? | |
Status: | Closed/Complete | Components: | Applications/app_dial Applications/app_exec pjproject/pjsip Resources/res_agi |
Versions: | 18.12.0 | Frequency of Occurrence | |
Related Issues: | |||
Environment: | ubuntu 20 | Attachments: | |
Description: | I need to send SIP headers to the outgoing channel so that they are present in the INVITE. I need to be "notified" if the channel answers the call.
Using Asterisk 13 and cha_sip.c and FastAGI I do it like this: I use SIPAddHeader to send the header and Macro in the Dial command to be "notfied" if the call is answered. This mechanism works on that version and before. (13) I just upgraded to asterisk 18 and now I am using PJSIP since chan_sip is deprecated and I also see that Macros are also deprecated and we should use pre-dial handlers and GoSub The problem I have is that if I use both in the DIAL command only the pre-dial handler is executed, the GoSub is ignored. If I omit the pre-dial handler the GoSub works this works: (only GoSub) EXEC DIAL PJSIP/5874513051231234@MYPBXOut,60,U(subConnect^agi://192.168.167.103:30075/ConnectionEstablished^364250168^PJSIP/3030-00000088^905^3051231234)L(7196000:75000)g and this DOES NOT work... only the pre-dial handler works, the GoSub is ignored EXEC DIAL PJSIP/5874513051231234@MYPBXOut,60,b(addHeaders^addheaderOutTrunkID^1(905^1657059780777001010)),U(subConnect^agi://192.168.167.103:30075/ConnectionEstablished^364250168^PJSIP/3030-00000088^905^3051231234)L(7196000:75000)g | ||
Comments: | By: Asterisk Team (asteriskteam) 2022-07-06 14:13:54.297-0500 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: Joshua C. Colp (jcolp) 2022-07-06 14:28:58.110-0500 This is not a bug. You have placed a "," in your list of options before the "U" option, causing the rest of the options to not be used. By: Fabian Borot (fborot) 2022-07-06 15:05:07.596-0500 Thank you Joshua.. indeed it works like that. we can close this By: Asterisk Team (asteriskteam) 2022-07-06 15:05:07.842-0500 This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable. |