| Summary: | ASTERISK-08543: G729 No path to translate. | ||
| Reporter: | David Faulk (dfaulk) | Labels: | |
| Date Opened: | 2007-01-10 09:55:12.000-0600 | Date Closed: | 2007-01-10 10:07:32.000-0600 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Core/CodecInterface |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | We have two Asterisk servers, one running 1.2 and one running 1.4 both connected to the same SIP VOIP provider. On the 1.2 server, until the call is answered we can see 0/0 encoders and decoders in use , that switches to 1/1 once th call is answered. During a call on the 1.4 server we see 1/1 encoders and decoders and while the call is ringing then we get this error: ~~~~ Call is ringing ~~~~ This is the 1.4 server. CLI> show g729 1/1 encoders/decoders of 1 licensed channels are currently in use [Jan 9 11:57:19] WARNING[25731]: rtp.c:874 ast_rtcp_read: RTCP Read too short [Jan 9 11:57:24] WARNING[25731]: rtp.c:874 ast_rtcp_read: RTCP Read too short ~~~~ Call is answered ~~~~ -- SIP/icallglobe-081f74f8 answered SIP/lukas-b774dc90 [Jan 9 11:57:24] WARNING[25731]: channel.c:3033 ast_channel_make_compatible: No path to translate from SIP/lukas-b774dc90(524290) to SIP/icallglobe-081f74f8(256) [Jan 9 11:57:24] WARNING[25731]: app_dial.c:1592 dial_exec_full: Had to drop call because I couldn't make SIP/lukas-b774dc90 compatible with SIP/icallglobe-081f74f8 | ||
| Comments: | By: Joshua C. Colp (jcolp) 2007-01-10 10:07:31.000-0600 Duplicate of 8781. | ||