| Summary: | ASTERISK-29451: RTP IP issue, Reverse Proxy using firewalld and OpenVPN | ||
| Reporter: | Alireza Mortazavi (morty71) | Labels: | |
| Date Opened: | 2021-05-25 14:15:40 | Date Closed: | 2021-06-09 12:00:00 |
| Priority: | Minor | Regression? | |
| Status: | Closed/Complete | Components: | Channels/chan_pjsip Channels/chan_sip/General |
| Versions: | 16.15.0 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | Linux raspbx 5.10.17-v7l+ | Attachments: | |
| Description: | After opening a community thread in freepbx forum and creating an issue in freepbx Jira, freepbx staffs asked me to create an issue here, because it could be an asterisk bug.
For details, please check this community topic, it includes the problem, logs, etc https://community.freepbx.org/t/rtp-issue-reverse-proxy-using-firewalld-and-openvpn/74127 In the logs, you can see that the public IP has been properly substituted in the signaling, but it hasn't been changed for the media. | ||
| Comments: | By: Asterisk Team (asteriskteam) 2021-05-25 14:15:41.204-0500 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: Asterisk Team (asteriskteam) 2021-05-25 14:15:41.818-0500 The severity of this issue has been automatically downgraded from "Blocker" to "Major". The "Blocker" severity is reserved for issues which have been determined to block the next release of Asterisk. This severity can only be set by privileged users. If this issue is deemed to block the next release it will be updated accordingly during the triage process. By: Joshua C. Colp (jcolp) 2021-05-25 14:26:00.305-0500 Thanks for the report and debug. However we also need protocol specific debug captured at the time of the issue. Please include the following: * Asterisk log files generated using the instructions on the Asterisk wiki [1], with the appropriate protocol debug options enabled, e.g. 'pjsip set logger on' if the issue involves the chan_pjsip channel driver. * Configuration information for the relevant channel driver, e.g. pjsip.conf. * A wireshark compatible packet capture, captured at the same time as the Asterisk log output. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information By: Joshua C. Colp (jcolp) 2021-05-25 14:27:33.392-0500 Additionally please attach the complete PJSIP configuration, including transports AND endpoint. By: Alireza Mortazavi (morty71) 2021-05-25 14:30:50.129-0500 Hi Joshua, the logs are mentioned in the freepbx community thread, I just place the links to the pastebin here too http://pastebin.freepbx.org/view/f8c62d61 http://pastebin.freepbx.org/view/eca67f45 my /etc/asterisk/pjsip.transports.conf [0.0.0.0-udp] type=transport protocol=udp bind=0.0.0.0:5060 external_media_address=1.2.3.4 external_signaling_address=1.2.3.4 allow_reload=no tos=cs3 cos=3 local_net=192.168.1.0/24 /etc/asterisk/pjsip.transports_custom.conf and /etc/asterisk/psjip.transports_custom_post.conf are empty By: Joshua C. Colp (jcolp) 2021-05-25 14:34:03.460-0500 Those logs contain SIP traces, but do not contain Asterisk debug logging as stated in the linked wiki page. Additionally you have not provided the endpoint definition as well. All information should also be attached so it does not rely on external sites or places to be available. By: Asterisk Team (asteriskteam) 2021-06-09 12:00:00.524-0500 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines | ||