jrhopper@pasty.com
2003-Nov-06 23:20 UTC
[Asterisk-Users] this is the code that breaks outgoing calls on grandstream
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
Mark Spencer
2003-Nov-07 07:47 UTC
[Asterisk-Users] this is the code that breaks outgoing calls on grandstream
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@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 >
jrhopper@pasty.com
2003-Nov-07 11:31 UTC
[Asterisk-Users] this is the code that breaks outgoing calls on grandstream
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@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@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 > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- A non-text attachment was scrubbed... Name: sip_debug Type: application/octet-stream Size: 12367 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20031107/606704b7/sip_debug.obj