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Merged revisions 103385 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103385 | file | 2008-02-12 11:09:24 -0400 (Tue, 12 Feb 2008) | 4 lines Even if no CallerID name or number has been provided by the remote party still use the configured sip.conf ones. (closes issue #11977) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -11004,13 +11004,13 @@ static enum check_auth_result check_user_ok(struct sip_pvt *p, char *of,
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ast_set_flag(&p->flags[0], SIP_CALL_LIMIT);
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if (!ast_strlen_zero(user->context))
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ast_string_field_set(p, context, user->context);
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if (!ast_strlen_zero(user->cid_num) && !ast_strlen_zero(p->cid_num)) {
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if (!ast_strlen_zero(user->cid_num)) {
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char *tmp = ast_strdupa(user->cid_num);
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if (ast_is_shrinkable_phonenumber(tmp))
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ast_shrink_phone_number(tmp);
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ast_string_field_set(p, cid_num, tmp);
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}
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if (!ast_strlen_zero(user->cid_name) && !ast_strlen_zero(p->cid_num))
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if (!ast_strlen_zero(user->cid_name))
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ast_string_field_set(p, cid_name, user->cid_name);
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ast_string_field_set(p, username, user->name);
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ast_string_field_set(p, peername, user->name);
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@@ -11147,13 +11147,13 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of,
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/* XXX this takes the name from the caller... can we override ? */
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ast_string_field_set(p, authname, peer->username);
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}
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if (!ast_strlen_zero(peer->cid_num) && !ast_strlen_zero(p->cid_num)) {
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if (!ast_strlen_zero(peer->cid_num)) {
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char *tmp = ast_strdupa(peer->cid_num);
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if (ast_is_shrinkable_phonenumber(tmp))
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ast_shrink_phone_number(tmp);
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ast_string_field_set(p, cid_num, tmp);
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}
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if (!ast_strlen_zero(peer->cid_name) && !ast_strlen_zero(p->cid_name))
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if (!ast_strlen_zero(peer->cid_name))
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ast_string_field_set(p, cid_name, peer->cid_name);
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ast_string_field_set(p, fullcontact, peer->fullcontact);
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if (!ast_strlen_zero(peer->context))
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