[Asterisk-Users] this is the code that breaks outgoing calls on grandstream

[jrhopper at pasty.com]

The broken code sends audio directly to the NAT address. For instance:

When I place a call from the grandstream to * with the broken code
* sees the grandstream. 
* according to tcpdump, sends a bunch of UDP (audio?) directly to the PRIVATE IP (192.168.0.100) of the grandstream even though it is on the other side of the internet and blocked by our border routers.

The code I commented out in the awful patch that I too hastily submitted last night.
Fixed calls from grandstream to *
Broke calls from * to grandstream in the same way they had been broken on incoming.

Attached is my sip_debug session for a failed call from my grandstream (192.168.0.100) through my cable modem (12.210.107.234) to my asterisk box (198.70.xx.xxx).

Jon


Mark Spencer <markster at digium.com> wrote ..
> There would have to be a corresponding change in the SIP dialog or in the
> actual audio sent both ways.  Can you provide some information on how it
> has changed?
> 
> Mark
> 
> On Fri, 7 Nov 2003 jrhopper at pasty.com wrote:
> 
> > Here is the diff from chan_sip.c 15 days ago and 16 days ago. 15 days
> ago is the point outgoing calls made via grandstream budgetone stopped
> working.
> >
> > Any help on why it breaks? Any possible fix?
> >
> > /tmp# diff asterisk/channels/chan_sip.c asterisk.works/channels/chan_sip.c
> > 289d288
> > <       int capability;
> > 3921,3922d3919
> > <                               p->capability = user->capability;
> > <                               p->jointcapability = user->capability;
> > 3963,3964d3959
> > <                               p->capability = peer->capability;
> > <                               p->jointcapability = peer->capability;
> > 5636d5630
> > <               user->capability = capability;
> > 5698,5709d5691
> > <                       } else if (!strcasecmp(v->name, "allow")) {
> > <                               format = ast_getformatbyname(v->value);
> > <                               if (format < 1)
> > <                                       ast_log(LOG_WARNING, "Cannot
> allow unknown format '%s'\n", v->value);
> > <                               else
> > <                                       user->capability |= format;
> > <                       } else if (!strcasecmp(v->name, "disallow"))
> {
> > <                               format = ast_getformatbyname(v->value);
> > <                               if (format < 1)
> > <                                       ast_log(LOG_WARNING, "Cannot
> disallow unknown format '%s'\n", v->value);
> > <                               else
> > <                                       user->capability &= ~format;
> > 5852,5855d5833
> > <                       } else if (!strcasecmp(v->name, "callgroup"))
> {
> > <                               peer->callgroup = ast_get_group(v->value);
> > <                       } else if (!strcasecmp(v->name, "pickupgroup"))
> {
> > <                               peer->pickupgroup = ast_get_group(v->value);
> > 5861a5840,5843
> > >                       } else if (!strcasecmp(v->name, "callgroup"))
> {
> > >                               peer->callgroup = ast_get_group(v->value);
> > >                       } else if (!strcasecmp(v->name, "pickupgroup"))
> {
> > >                               peer->pickupgroup = ast_get_group(v->value);
> >
> > Jon
> >
> 
> _______________________________________________
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> Asterisk-Users at lists.digium.com
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