Summary: | ASTERISK-25568: [patch]180 Ringing not sent after 183 Session Progress | ||
Reporter: | Morten Tryfoss (mtryfoss) | Labels: | patch |
Date Opened: | 2015-11-18 06:54:10.000-0600 | Date Closed: | |
Priority: | Minor | Regression? | |
Status: | Open/New | Components: | Channels/chan_sip/General |
Versions: | 13.6.0 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | Attachments: | ( 0) chan_sip.patch | |
Description: | We've got a simple scenario with a SIP call through Asterisk.
sip.conf progressinband=no (default setting) The remote switch sends a 100 Trying and a 183 Session Progress immediately after we send the INVITE. When the remote destination starts ringing (CPG received on ISUP-side of the switch), the switch sends a 180 Ringing to Asterisk. This is not forwarded to the other side, which I think is wrong. This can cause lack of ringback tone or wrongly mapped release causes in the originating switch. For example, a 480 temp unavailable is mapped to "20 - subscriber absent" if it has not started to ring, or "19 - no answer from user" if it has. The attached patch solves the problem for me. | ||
Comments: | By: Asterisk Team (asteriskteam) 2015-11-18 06:54:12.456-0600 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. 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]. By: Morten Tryfoss (mtryfoss) 2015-11-18 06:54:38.313-0600 Attached patch. By: Olle Johansson (oej) 2015-11-18 06:59:21.422-0600 I think the assumption that we should not move to ringing state when early media is already active is wrong. This seems to be copied to a few other places in chan_sip. Early media is independent of ringing. By: Morten Tryfoss (mtryfoss) 2015-11-19 01:42:38.148-0600 What feedback do you need? By: Rusty Newton (rnewton) 2015-11-19 17:56:01.199-0600 Sorry I thought I had commented with a canned response to remind you about getting the patch on Gerrit: Once you've followed the Code Review process [1] and submitted your code to Gerrit [2] be sure to edit this JIRA issue and add the Gerrit review URL in the appropriate field. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Code+Review [2] https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage |