Merged revisions 62989 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62989 | file | 2007-05-03 13:44:00 -0300 (Thu, 03 May 2007) | 10 lines Merged revisions 62987 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2 lines When a peer is seeded or built tell the devicestate core to update it's status. This is easier then having chan_sip load before pbx_config. (issue #9658 reported by dlynes) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@ -8256,6 +8256,7 @@ static void reg_source_db(struct sip_peer *peer)
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ast_sched_del(sched, peer->expire);
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peer->expire = ast_sched_add(sched, (expiry + 10) * 1000, expire_register, peer);
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register_peer_exten(peer, TRUE);
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ast_device_state_changed("SIP/%s", peer->name);
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}
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/*! \brief Save contact header for 200 OK on INVITE */
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@ -17419,6 +17420,7 @@ static int reload_config(enum channelreloadreason reason)
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if (ast_true(hassip) || (!hassip && genhassip)) {
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peer = build_peer(cat, gen, ast_variable_browse(ucfg, cat), 0);
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if (peer) {
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ast_device_state_changed("SIP/%s", peer->name);
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ASTOBJ_CONTAINER_LINK(&peerl,peer);
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unref_peer(peer);
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peer_count++;
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