| Summary: | ASTERISK-07506: I mad changes in the channels SIP and OH323 | ||
| Reporter: | Thiago Maluf (malufrj) | Labels: | |
| Date Opened: | 2006-08-10 07:30:43 | Date Closed: | 2006-08-10 11:42:30 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | I am student and work in the VoIP project in my university. I was need that the variables SIPCALLID and H323CONF_ID be save when Asterisk make call. For example, when the Asterisk receive one call by channel SIP, it save the variable SIPCALLID, but when it make call to channel SIP, it don`t save. The same thing happened in the Channel OH323. After, I had studing the channels, I get mad a code that fix it. And I would like send it for you to put in the new version of Asterisk. If possible, I would like participate the chats with the developers of Asterisk, and if possible, be a developer of Asterisk. I had studing much the Asterisk and I want in the next year make the Asterisk Certification. Very thanks for advanced, Thiago Maluf Resende. | ||
| Comments: | By: Serge Vecher (serge-v) 2006-08-10 08:59:09 Ok, sounds great. Here is some homework for you. 1. Read the bug posting guidelines http://www.digium.com/bugguidelines.html 2. Get a disclaimer on file (see bottom of http://bugs.digium.com/main_page.php) and cofirm with a note here when done. 3. Upload the changes you've made as a patchfile. See http://www.asterisk.org/developers/Patch_Howto 4. If you would like to start with Asterisk development, please take a look at the open bugs on this bug-tracker and see which issues you can tackle by yourself. By: Clod Patry (junky) 2006-08-10 11:42:29 also, check the mailing list asterisk-dev and #asterisk-dev on IRC (freenode). | ||