[Asterisk-Dev] chan_misdn in asterisk beta 2 ?
Zoa
zoachien at securax.org
Tue Nov 1 11:45:11 MST 2005
We are currently not working on it, we have something more urgent to do,
when that is over, we will get back to the new sip jb implementation.
If it can be made to not be compiled by default (and nothing changes by
default) then i might agree to have it in the 1.2, but if it cannot
completely be ifdef'd then i'd say leave it out.
Zoa
Ray Van Dolson wrote:
>On Tue, Nov 01, 2005 at 12:09:27PM -0600, Kevin P. Fleming wrote:
>
>
>>Eric "ManxPower" Wieling wrote:
>>
>>
>>
>>>I think it's much more important to get a SIP jitterbuffer into 1.2.
>>>
>>>
>>That's still on the table, it might just happen :-)
>>
>>
>
>Any plans to get this into CVS-HEAD for testing? I haven't seen any activity
>on the "bug report" in a while. The current incarnation of the patch works
>fine for SIP ATA -> Asterisk -> PSTN but fails miserably with SIP ATA ->
>Asterisk -> AudioCodes SIP GW -> PSTN.
>
>Maybe it's the AudioCodes, but it definitely doesn't sound very good :)
>
>Ray
>_______________________________________________
>Asterisk-Dev mailing list
>Asterisk-Dev at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-dev
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 254 bytes
Desc: OpenPGP digital signature
Url : http://lists.digium.com/pipermail/asterisk-dev/attachments/20051101/b7cd109e/signature.pgp
More information about the asterisk-dev
mailing list