| Summary: | ASTERISK-09998: When I allow the codec G729 first in sip.conf the other codecs are not offered in the INVITE | ||
| Reporter: | Antonio Latorre (antxoneti) | Labels: | |
| Date Opened: | 2007-08-01 01:17:52 | Date Closed: | 2011-06-07 14:07:49 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Codecs/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | I have made to test: 1- sip.conf disallow = all allow = g729 allow = alaw allow = ulaw And the trace of the INVITE is m=audio 12480 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. No more codecs are offered than the G729 2- sip.conf disallow = all allow = alaw allow = ulaw allow = g729 And the trace INVITE is: m=audio 19360 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. The codec G729 is not offered ****** ADDITIONAL INFORMATION ****** I have made the test with the version 1.4.4 copying the same configuration files and changing the order and I had the three codecs | ||
| Comments: | By: Joshua C. Colp (jcolp) 2007-08-01 07:46:42 Please provide a full SIP debug when reporting SIP related bugs, and also a description of the call flow. By: Joshua C. Colp (jcolp) 2007-08-16 13:54:26 I have tried again to reproduce this without any luck. Since the needed information has yet to be provided and it's been 2 weeks I'm suspending this for now. | ||