[Asterisk-Users] this is the code that breaks outgoing calls on
grandstream
Mark Spencer
markster at digium.com
Fri Nov 7 07:47:42 MST 2003
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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