[Asterisk-Dev] mgcp<>sip dirty fix

[Michael Baird]

Alex what's the symptoms this fix cure, I'm having problems with mgcp ->
sip as well, my mgcp phone will initiate a call behind asterisk travel
sip to my cisco pstn gateway, I get one ring and it drops basically.
MGCP -> MGCP works fine, and zap -> mgcp, and mgcp -> zap work fine, sip
-> zap, and zap -> sip seem to be fine as well.

Regards
MIKE


On Fri, 2003-03-28 at 07:53, alex at pilosoft.com wrote:
> Like above, this is not a correct fix, but it'll allow sip<>mgcp 
> interoperation. I don't know enough to make a proper fix.
> 
> Index: channels/chan_sip.c
> ===================================================================
> RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
> retrieving revision 1.15
> diff --unified -r1.15 chan_sip.c
> --- channels/chan_sip.c 28 Mar 2003 06:59:34 -0000 1.15
> +++ channels/chan_sip.c 28 Mar 2003 13:06:35 -0000
> @@ -4460,8 +4468,10 @@
>         struct sip_pvt *p;
>         p = chan->pvt->pvt;
>         if (p) {
> +/* this breaks SIP<>MGCP 
>                 p->outgoing = 1;
>                 transmit_reinvite_with_sdp(p, rtp);
> +*/
>                 return 0;
>         }
> 
> 
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