| Summary: | ASTERISK-30356: suspect ps_endpints option rewrite_contact not rewriting with source ip. | ||
| Reporter: | Michael (ringo) | Labels: | |
| Date Opened: | 2022-12-15 10:48:52.000-0600 | Date Closed: | 2022-12-15 11:22:55.000-0600 |
| Priority: | Minor | Regression? | |
| Status: | Closed/Complete | Components: | pjproject/pjsip Resources/res_rtp_asterisk |
| Versions: | 20.0.1 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | debian 10 amd64 | Attachments: | |
| Description: | I suspect asterisk and/or pjsip is not rewriting the contact header with source ip address and port when using the rewrite_contact option in ps_endpoints.
I have used this option in the past with success, when address is an internal nat'd 192.168 address. Recently, using soft phone on cellular data network, i have seen ipv6 ip addresses show up in the sip headers while the source address is indeed ipv4. SIP packets move back and forth but when answering a call, no audio is passed and asterisk console shows this: WARNING[4801] acl.c: Cannot connect to 2605:b100:349:5cd1:0:4:375a:3e01: Cannot assign requested address it looks like the outbound connection is trying to go to an ipv6 address instead of the source ipv4 address. | ||
| Comments: | By: Asterisk Team (asteriskteam) 2022-12-15 10:48:55.990-0600 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) 2022-12-15 10:53:36.393-0600 The rewrite_contact option has no affect on audio. The rtp_symmetric option sends audio to where it is received from. There is also no "outbound connection". I'd suggest seeking help on the community forum first[1] with logs and information, after which if an issue is determined then this can be reopened. [1] https://community.asterisk.org/ By: Michael (ringo) 2022-12-15 11:20:03.198-0600 ok thanks | ||