| Summary: | ASTERISK-10040: SIP_CODEC variable does not change the codec | ||
| Reporter: | andykwg (andykwg) | Labels: | |
| Date Opened: | 2007-08-07 22:56:12 | Date Closed: | 2011-06-07 14:08:12 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/CodecHandling |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | I need to use get the codec type from a DB/mysql and set the codec right before using the dial command. There is the function try_suggested_sip_code which seems to pick up the SIP_CODEC variable and set the codec. But it does not seems to be setting the codec as per requested. I try to use the static client setup within sip.conf and codec selection works there. But SIP_CODEC does not change the codec at dial time. ****** ADDITIONAL INFORMATION ****** I just try a simple dialplan as follows: exten=ad,1,Set(__SIP_CODEC=g729) exten=ad,n,Dial(${CALLURI},${CALLRING},${CALLOPTION}) In the sip.conf, I have allowed g729, alaw and ulaw. | ||
| Comments: | By: Joshua C. Colp (jcolp) 2007-08-08 07:38:22 I'm suspending this bug since it appears to be a feature request. I've confirmed that SIP_CODEC has never changed the codec on the outgoing call, just the incoming one. I have also confirmed that it works fine and dandy. If this is not the case feel free to reopen. | ||