diff options
| author | Matthew Jordan <mjordan@digium.com> | 2015-02-12 20:34:37 +0000 |
|---|---|---|
| committer | Matthew Jordan <mjordan@digium.com> | 2015-02-12 20:34:37 +0000 |
| commit | 29f66b0429f4314e082bebcf0630b016b317cba3 (patch) | |
| tree | ef1ca239afe19ed5ac43cee1bd46c4678393dd21 /res/res_pjsip_nat.c | |
| parent | 9d081ed06cc32380d541ce4cb317bc23c32dee56 (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_nat.c')
| -rw-r--r-- | res/res_pjsip_nat.c | 3 |
1 files changed, 3 insertions, 0 deletions
diff --git a/res/res_pjsip_nat.c b/res/res_pjsip_nat.c index 588734352..b71a84bbd 100644 --- a/res/res_pjsip_nat.c +++ b/res/res_pjsip_nat.c @@ -52,6 +52,8 @@ static pj_bool_t handle_rx_message(struct ast_sip_endpoint *endpoint, pjsip_rx_d uri->transport_param.slen = 0; } 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); /* rewrite the session target since it may have already been pulled from the contact header */ if (dlg && (!dlg->remote.contact @@ -205,6 +207,7 @@ static pj_status_t nat_on_tx_message(pjsip_tx_data *tdata) pj_strdup2(tdata->pool, &uri->host, ast_sockaddr_stringify_host(&transport->external_address)); if (transport->external_signaling_port) { uri->port = transport->external_signaling_port; + ast_debug(4, "Re-wrote Contact URI port to %d\n", uri->port); } } |