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Summary:ASTERISK-27785: Hangup after generate call via manager with EarlyMedia
Reporter:adilson leffa magnus (magnusbilling)Labels:pjsip
Date Opened:2018-04-02 09:25:17Date Closed:2018-04-04 14:25:27
Priority:MajorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:15.3.0 Frequency of
Occurrence		Constant
Related
Issues:
Environment:Centos 7 on google cloudAttachments:
Description:Hello.

Generate a call via a socket with EarlyMedia: yes

when receive SIP/2.0 183 Ringing. the call bridge on context. But after this moment is impossible to hangup the call.

I try:

Execute hangup in the context
Execute on Asterisk console the command hangup request SIP/XXXX-00000006
wait Timeout

In all cases Asterisk not send the CANCEL packet.
Include I can stop Asterisk server and the call continues ringing on the phone until answer.


If i execute the hangup before receive the SIP/2.0 183 Ringing it work ok.

If i not user EarlyMedia, the hangup via "hangup request CHANNEL" work ok too, or execute "hangup request CHANNEL" before receive the 183.

The context that I bridge the call is very simples
exten => 1,Hangup(); But i can try anything and the same problem occurs.



$socket = fsockopen('127.0.0.1', '5038', $errno, $errstr, 5);
                   fputs($socket, "Action: Login\r\n");
                   fputs($socket, "UserName: admin\r\n");
                   fputs($socket, "Secret: pass\r\n\r\n");
                   fputs($socket, "Action: Originate\r\n");
                   fputs($socket, "Channel: SIP/xxxxxxxx@trunk\r\n");
                   fputs($socket, "Exten: 1000\r\n");
                   fputs($socket, "Context: billing\r\n");
                   fputs($socket, "Timeout: 12000\r\n");
                   fputs($socket, "Async: yes\r\n");
                   fputs($socket, "EarlyMedia: true\r\n");
                   fputs($socket, "Priority: 1\r\n");
                   fputs($socket, "Action: Logoff\r\n\r\n");



I try Asterisk 13, 14 and 15

Comments:By: Asterisk Team (asteriskteam) 2018-04-02 09:25:18.500-0500

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].
By: Joshua C. Colp (jcolp) 2018-04-03 07:18:32.597-0500

Have you tried this with PJSIP to see if it works as expected?

It's likely isolated to chan_sip, which is under community support so there is no timeframe on when it would get looked into if this is the case.
By: adilson leffa magnus (magnusbilling) 2018-04-03 10:24:40.350-0500

i'll try with pjsip and fedback for you.
By: adilson leffa magnus (magnusbilling) 2018-04-04 14:24:35.661-0500

hi, I test witj PJSIP and work perfectly. Thanks

By: adilson leffa magnus (magnusbilling) 2018-04-04 14:25:27.239-0500

the same project work with PJSIP. But the problem really exists in chan_sip