[Asterisk-Users] more about SIP ...

[Martin Pycko]

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 at 216.52.153.207 : Go2Call SIP gateway
>
>
>
>     -- Executing Dial("SIP/217.168.168.49:5060", "SIP/723 at 216.52.153.207")
> in new stack
>     -- Called 723 at 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
>
>
>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>