| Summary: | ASTERISK-29028: Queue not ring PJSIP extension | ||
| Reporter: | Roberto (tel.medola) | Labels: | |
| Date Opened: | 2020-08-13 21:33:12 | Date Closed: | 2020-08-14 17:02:22 |
| Priority: | Minor | Regression? | |
| Status: | Closed/Complete | Components: | Applications/app_queue |
| Versions: | 17.6.0 | Frequency of Occurrence | One Time |
| Related Issues: | |||
| Environment: | Centos 7.6 PJSIP 2.10 Asterisk with ARA - PostgreSQL 11 | Attachments: | |
| Description: | I am having a lot of problems with SIP through NAT. So, I decided to adopt PJSIP. However, I am not able to make the extensions ring when receiving a call from the queue. I'm using telnet to include the extension and on the asterisk console, it even shows Called PJSIP/6001, but the extension doesn't ring. If I call from extension to extension, it works normally.
telenet: Action: QueueAdd Queue: queuetest MemberName: 1234 Interface: PJSIP/6001 StateInterface: PJSIP/6001 Ringinuse: yes Paused: false If I change to SIP, the extension will call normally. My configuration pjsip.conf [transport-udp-nat] type=transport protocol=udp bind=0.0.0.0:5160 local_net=192.0.0.0/24 external_media_address=192.168.0.196 external_signaling_address=192.168.0.196 [6001] type=endpoint context=callcenter disallow=all allow=g729 allow=ulaw allow=gsm auth=6001 aors=6001 transport=transport-udp-nat direct_media=no allow_subscribe=yes sub_min_expiry=30 [6001] type=auth auth_type=userpass password=6001 username=6001 [6001] type=aor max_contacts=99 Thanks! | ||
| Comments: | By: Asterisk Team (asteriskteam) 2020-08-13 21:33:13.125-0500 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. By: Kevin Harwell (kharwell) 2020-08-14 17:01:49.269-0500 We appreciate the difficulties you are facing, however this does not appear to be a bug report and your request or comments would be better served in a different forum. The Asterisk community provides support over IRC, mailing lists, and forums [1] as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors [2]. Thanks! [1] https://community.asterisk.org/ [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines | ||