| Summary: | ASTERISK-27085: [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip | ||
| Reporter: | Torrey Searle (tsearle) | Labels: | pjsip |
| Date Opened: | 2017-06-26 07:51:37 | Date Closed: | 2017-08-16 06:45:18 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | Channels/chan_pjsip |
| Versions: | 13.15.1 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | SIPDtmfMode allowed you to change the dtmf mode on a per-call basis. This is useful when the adjacent peer / endpoint is a kamailio that is forwarding traffic from a variety of different types of equipment
This patch introduces the function PJSIPDtmfMode that provides the same functionality to PJSSIP | ||
| Comments: | By: Asterisk Team (asteriskteam) 2017-06-26 07:51:38.449-0500 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. 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]. | ||