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Merged revisions 48168 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48168 | file | 2006-11-30 16:18:24 -0500 (Thu, 30 Nov 2006) | 2 lines Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -163,7 +163,6 @@ struct gtalk_container {
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};
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static const char desc[] = "Gtalk Channel";
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static const char type[] = "Gtalk";
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static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
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@@ -192,7 +191,7 @@ static int gtalk_get_codec(struct ast_channel *chan);
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/*! \brief PBX interface structure for channel registration */
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static const struct ast_channel_tech gtalk_tech = {
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.type = type,
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.type = "Gtalk",
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.description = "Gtalk Channel Driver",
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.capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
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.requester = gtalk_request,
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@@ -220,7 +219,7 @@ static struct in_addr __ourip;
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/*! \brief RTP driver interface */
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static struct ast_rtp_protocol gtalk_rtp = {
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type: "gtalk",
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type: "Gtalk",
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get_rtp_info: gtalk_get_rtp_peer,
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set_rtp_peer: gtalk_set_rtp_peer,
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get_codec: gtalk_get_codec,
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@@ -921,10 +920,12 @@ static struct ast_channel *gtalk_new(struct gtalk *client, struct gtalk_pvt *i,
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fmt = ast_best_codec(tmp->nativeformats);
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if (i->rtp) {
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ast_rtp_setstun(i->rtp, 1);
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tmp->fds[0] = ast_rtp_fd(i->rtp);
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tmp->fds[1] = ast_rtcp_fd(i->rtp);
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}
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if (i->vrtp) {
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ast_rtp_setstun(i->rtp, 1);
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tmp->fds[2] = ast_rtp_fd(i->vrtp);
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tmp->fds[3] = ast_rtcp_fd(i->vrtp);
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}
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@@ -1790,7 +1791,7 @@ static int load_module(void)
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/* Make sure we can register our channel type */
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if (ast_channel_register(>alk_tech)) {
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ast_log(LOG_ERROR, "Unable to register channel class %s\n", type);
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ast_log(LOG_ERROR, "Unable to register channel class %s\n", gtalk_tech.type);
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return -1;
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}
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return 0;
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