18 dec 2007 kl. 06.50 skrev SVN commits to the Digium repositories:
> Author: murf
> Date: Mon Dec 17 23:50:15 2007
> New Revision: 93535
>
> URL: http://svn.digium.com/view/asterisk?view=rev&rev=93535
> Log:
> Good. the dialog callids change, hadn't known that. Now I do. Unlink
> and relink in the hash table under new name. Sip works better now.
No, they don't.
You've ropably discovered a bug. When, exactly does this happen? Do
you have a call trace (sip debug) from Asterisk so we
can fix it ... ???
Guess we need to find each other on IRC 
/O
It looks pretty purposeful to me. Albeit, I realize that some of these freshly create a new dialog, and then set the name; this still qualifies as a name change (to me)-- although it might not to you. Here's my enumeration of the places:
In create_addr_from_peer() :
if (!ast_strlen_zero(peer->fromdomain)) {
ast_string_field_set(dialog, fromdomain, peer->fromdomain);
if (!dialog->initreq.headers) {
char *c;
char *tmpcall = ast_strdupa(dialog->callid);
/* this sure looks to me like we are going to change the callid on this dialog!! */
c = strchr(tmpcall, ' ');
if (c) {
*c = ' ';
ao2_unlink(dialogs,dialog,"About to change the callid -- remove the old name");
ast_string_field_build(dialog, callid, "%s%s", tmpcall, peer->fromdomain);
ao2_link(dialogs,dialog,"New dialog callid -- inserted back into table";);
ast_log(LOG_NOTICE,"=========Changed Dialog name to %sn", dialog->;callid);
}
}
}
Note: see the ast_string_field_build() call above?
Note2: I'm showing "fixed" code, so in the original, there's no ao2_unlink, ao2_link calls (of course, since in the original,
astobj2 isn't being used!)
In sip_notify() :
/* Recalculate our side, and recalculate Call ID */
ast_sip_ouraddrfor(&p->sa.sin_addr, &p->;ourip);
build_via(p);
ast_log(LOG_NOTICE,";============About to build callid for %sn", p->callid);
ao2_unlink(dialogs,p,"About to change the callid -- remove the old name");
build_callid_pvt(p);
ao2_link(dialogs,p,"Linking in new name");
ast_log(LOG_NOTICE,"================New callid: %sn", p->callid);
ast_cli(a->;fd, "Sending NOTIFY of type '%s' to '%s'n";, a->argv[2], a->argv[i]);
dialog_ref(p,"bump the count of p, which transmit_sip_request will decrement.");
transmit_sip_request(p, &req);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
dialog_unref(p,"unref pvt at end of for loop in sip_notify");
Note: see the call to build_callid_pvt() above...
In sip_send_mwi_to_peer():
if (peer->mwipvt) {
...
} else {
/* Build temporary dialog for this message */
if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY)))
return -1;
if (create_addr_from_peer(p, peer)) {
; ; ; ...
return 0;
}
/* Recalculate our side, and recalculate Call ID */
ast_sip_ouraddrfor(&p->sa.sin_addr, &p->;ourip);
ast_log(LOG_NOTICE,"=======================About to build callid for %sn", p->callid);
build_via(p);
ao2_unlink(dialogs,p,"About to change the callid -- remove the old name");
build_callid_pvt(p);
ao2_link(dialogs,p,"Linking in under new name");
ast_log(LOG_NOTICE,"====================New callid: %sn", p->callid);
/* Destroy this session after 32 secs */
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
Note: this one is pretty active, as it generates a new dialog, which sets up a name, and then create_addr_from_peer might modify it, and then we do a build_callid_pvt to finish up the process!
sip_poke_peer():
if (!(p = sip_alloc(NULL, NULL, 0, SIP_OPTIONS)))
return -1;
peer->call = dialog_ref(p,"copy sip alloc from p to peer->call");
...
/* Recalculate our side, and recalculate Call ID */
ast_log(LOG_NOTICE,"===============About to build callid for %sn", p->callid);
ast_sip_ouraddrfor(&p->sa.sin_addr, &p->;ourip);
build_via(p);
ao2_unlink(dialogs,p,";About to change the callid -- remove the old name");
build_callid_pvt(p);
ao2_link(dialogs,p,"Linking in under new name");
ast_log(LOG_NOTICE,"==============New callid: %sn", p->callid);
Note: again, this routine generates a new dialog via sip_alloc, which sets the dialog name, and then modifies it...
which in some places might not be classified as a name change, really, but **I** have consider it such.
sip_alloc() is called in find_call(), transmit_register(), sip_notify(), sip_send_mwi_to_peer(), sip_poke_peer(), and sip_request_call(), (The names in red are mentioned above). So, the routines above are not the only ones to allocate a dialog, and not all functions that allocate a dialog reset the name.
murf
--
Steve Murphy
Software Developer
Digium
|
|
