summaryrefslogtreecommitdiff
path: root/res/res_pjsip_transport_websocket.c
diff options
context:
space:
mode:
authorMatthew Jordan <mjordan@digium.com>2015-02-12 20:34:37 +0000
committerMatthew Jordan <mjordan@digium.com>2015-02-12 20:34:37 +0000
commit29f66b0429f4314e082bebcf0630b016b317cba3 (patch)
treeef1ca239afe19ed5ac43cee1bd46c4678393dd21 /res/res_pjsip_transport_websocket.c
parent9d081ed06cc32380d541ce4cb317bc23c32dee56 (diff)
ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app
This patch adds a new feature to ARI to redirect a channel to another server, and fixes a few bugs in PJSIP's handling of the Transfer dialplan application/ARI redirect capability. *New Feature* A new operation has been added to the ARI channels resource, redirect. With this, a channel in a Stasis application can be redirected to another endpoint of the same underlying channel technology. *Bug fixes* In the process of writing this new feature, two bugs were fixed in the PJSIP stack: (1) The existing .transfer channel callback had the limitation that it could only transfer channels to a SIP URI, i.e., you had to pass 'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is still supported, it is somewhat unintuitive - particularly in a world full of endpoints. As such, we now also support specifying the PJSIP endpoint to transfer to. (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by updating its Contact header. Alas, that resulted in the forwarding destination set by the dialplan application/ARI resource/whatever being rewritten with very incorrect information. Hence, we now don't bother updating an outgoing response if it is a 302. Since this took a looong time to find, some additional debug statements have been added to those modules that update the Contact headers. Review: https://reviewboard.asterisk.org/r/4316/ ASTERISK-24015 #close Reported by: Private Name ASTERISK-24703 #close Reported by: Matt Jordan ........ Merged revisions 431717 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res/res_pjsip_transport_websocket.c')
-rw-r--r--res/res_pjsip_transport_websocket.c2
1 files changed, 2 insertions, 0 deletions
diff --git a/res/res_pjsip_transport_websocket.c b/res/res_pjsip_transport_websocket.c
index 5616715e0..94902d65b 100644
--- a/res/res_pjsip_transport_websocket.c
+++ b/res/res_pjsip_transport_websocket.c
@@ -329,6 +329,8 @@ static pj_bool_t websocket_on_rx_msg(pjsip_rx_data *rdata)
pj_cstr(&uri->host, rdata->pkt_info.src_name);
uri->port = rdata->pkt_info.src_port;
+ ast_debug(4, "Re-wrote Contact URI host/port to %.*s:%d\n",
+ (int)pj_strlen(&uri->host), pj_strbuf(&uri->host), uri->port);
pj_strdup(rdata->tp_info.pool, &uri->transport_param, (type == (long)transport_type_ws) ? &STR_WS : &STR_WSS);
}