You need to have
disallow=all
allow=g723.1
and the other remote phone has to use also the G723 codec. Otherwise
asterisk will try to transcode but it doesn't have the G.723 code itself.
regards
Martin
On Fri, 6 Jun 2003, Dave Alan Caruana wrote:
> I added the line "allow G723.1" in my sip.conf general config,
> and from a bridge connection which gives silence,
> I have progressed to the error message below,
> and the call gets rejected.
>
> help!!
>
> Dave
>
> ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel
Instant
> Expressa
> 723@216.52.153.207 : Go2Call SIP gateway
>
>
>
> -- Executing Dial("SIP/217.168.168.49:5060",
"SIP/723@216.52.153.207")
> in new stack
> -- Called 723@216.52.153.207
> WARNING[1240577216]: File channel.c, Line 1711
> (ast_channel_make_compatible): No path to translate from
> SIP/216.52.153.207-2e12(1) to SIP/217.168.168.49:5060(4)
> -- SIP/216.52.153.207-2e12 answered SIP/217.168.168.49:5060
> WARNING[1240577216]: File channel.c, Line 1711
> (ast_channel_make_compatible): No path to translate from
> SIP/217.168.168.49:5060(4) to SIP/216.52.153.207-2e12(1)
> WARNING[1240577216]: File app_dial.c, Line 606 (dial_exec): Had to drop
call
> because I couldn't make SIP/217.168.168.49:5060 compatible with
> SIP/216.52.153.207-2e12
> == Spawn extension (default, 1303, 1) exited non-zero on
> 'SIP/217.168.168.49:5
>
>
>
>
>
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