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authorJoshua Colp <jcolp@digium.com>2017-05-09 10:25:29 +0000
committerJoshua Colp <jcolp@digium.com>2017-05-09 10:33:04 +0000
commit10a49ab3627b7e93c01b22376cdf71015422894c (patch)
treed74040ef408dcb338447f22f9b93c8ad1d4e8a01
parent371213217c8e2721fa06dedda578e3eb70270458 (diff)
res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages.
This change adds the required logic to allow the SIP Call-ID to be placed into the HEP RTCP traffic if the chan_sip module is used. In cases where the option is enabled but the channel is not either SIP or PJSIP then the code will fallback to the channel name as done previously. Based on the change on Nir's branch at: team/nirs/hep-chan-sip-support ASTERISK-26427 Change-Id: I09ffa5f6e2fdfd99ee999650ba4e0a7aad6dc40d
-rw-r--r--CHANGES6
-rw-r--r--configs/samples/hep.conf.sample6
-rw-r--r--res/res_hep_rtcp.c18
3 files changed, 25 insertions, 5 deletions
diff --git a/CHANGES b/CHANGES
index d93903dfe..990e5d8ac 100644
--- a/CHANGES
+++ b/CHANGES
@@ -33,6 +33,12 @@ res_pjsip_config_wizard
endpoint/outbound_proxy, aor/outbound_proxy and registration/outbound_proxy
parameters.
+res_hep_rtcp
+------------------
+ * If the 'call-id' value is specified for the uuid_type option and a
+ chan_sip channel is used the resulting HEP traffic will now contain the
+ SIP Call-ID instead of the Asterisk channel name.
+
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.14.0 to Asterisk 13.15.0 ----------
------------------------------------------------------------------------------
diff --git a/configs/samples/hep.conf.sample b/configs/samples/hep.conf.sample
index 3d1e74139..32bd8df39 100644
--- a/configs/samples/hep.conf.sample
+++ b/configs/samples/hep.conf.sample
@@ -24,5 +24,9 @@ capture_id = 1234 ; A unique integer identifier for this
; with each packet from this server.
uuid_type = call-id ; Specify the preferred source for the Homer
; correlation UUID. Valid options are:
- ; - 'call-id' for the PJSIP SIP Call-ID
+ ; - 'call-id' for the PJSIP or chan_sip SIP
+ ; Call-ID
; - 'channel' for the Asterisk channel name
+ ; Note: If 'call-id' is specified but the
+ ; channel is not PJSIP or chan_sip then the
+ ; Asterisk channel name will be used instead.
diff --git a/res/res_hep_rtcp.c b/res/res_hep_rtcp.c
index f4f1dfe3b..21e7d6be7 100644
--- a/res/res_hep_rtcp.c
+++ b/res/res_hep_rtcp.c
@@ -55,12 +55,22 @@ static char *assign_uuid(struct ast_json *json_channel)
return NULL;
}
- if (uuid_type == HEP_UUID_TYPE_CALL_ID && ast_begins_with(channel_name, "PJSIP")) {
- struct ast_channel *chan = ast_channel_get_by_name(channel_name);
+ if (uuid_type == HEP_UUID_TYPE_CALL_ID) {
+ struct ast_channel *chan = NULL;
char buf[128];
- if (chan && !ast_func_read(chan, "CHANNEL(pjsip,call-id)", buf, sizeof(buf))) {
- uuid = ast_strdup(buf);
+ if (ast_begins_with(channel_name, "PJSIP")) {
+ chan = ast_channel_get_by_name(channel_name);
+
+ if (chan && !ast_func_read(chan, "CHANNEL(pjsip,call-id)", buf, sizeof(buf))) {
+ uuid = ast_strdup(buf);
+ }
+ } else if (ast_begins_with(channel_name, "SIP")) {
+ chan = ast_channel_get_by_name(channel_name);
+
+ if (chan && !ast_func_read(chan, "SIP_HEADER(call-id)", buf, sizeof(buf))) {
+ uuid = ast_strdup(buf);
+ }
}
ast_channel_cleanup(chan);