Module: kamailio
Branch: master
Commit: 1151b0854f5c24ef3e9b7ee209efc31ce7ef0519
URL: https://github.com/kamailio/kamailio/commit/1151b0854f5c24ef3e9b7ee209efc31…
Author: �������������������� �������������������������������� <git-dpa(a)aegee.org>
Committer: Daniel-Constantin Mierla <miconda(a)gmail.com>
Date: 2023-04-10T08:05:56+02:00
cnxcc: typos
---
Modified: src/modules/cnxcc/doc/cnxcc_admin.xml
Modified: src/modules/cnxcc/example/kamailio-cnxcc.cfg
---
Diff: https://…
[View More]github.com/kamailio/kamailio/commit/1151b0854f5c24ef3e9b7ee209efc31…
Patch: https://github.com/kamailio/kamailio/commit/1151b0854f5c24ef3e9b7ee209efc31…
---
diff --git a/src/modules/cnxcc/doc/cnxcc_admin.xml b/src/modules/cnxcc/doc/cnxcc_admin.xml
index 322a227bfb7..2be0abdb541 100644
--- a/src/modules/cnxcc/doc/cnxcc_admin.xml
+++ b/src/modules/cnxcc/doc/cnxcc_admin.xml
@@ -450,7 +450,7 @@ if (!cnxcc_terminate_all("$var(customer)")) {
<section id="cnxcc.rpc.check_client">
<title>cnxcc.check_client</title>
<para>
- Retrives all calls from a particular identifier.
+ Retrieves all calls from a particular identifier.
</para>
<para>
Parameters: <emphasis>client/customer identifier</emphasis>
diff --git a/src/modules/cnxcc/example/kamailio-cnxcc.cfg b/src/modules/cnxcc/example/kamailio-cnxcc.cfg
index 9b11b26a8dc..236b2384902 100644
--- a/src/modules/cnxcc/example/kamailio-cnxcc.cfg
+++ b/src/modules/cnxcc/example/kamailio-cnxcc.cfg
@@ -6,12 +6,12 @@
#
# Kamailio SIP Server v3.2 - default configuration script
-# - web: http://www.kamailio.org
+# - web: https://www.kamailio.org
# - git: http://sip-router.org
#
# Direct your questions about this file to: <sr-users(a)lists.kamailio.org>
#
-# Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php
+# Refer to the Core CookBook at https://www.kamailio.org/w/documentation/
# for an explanation of possible statements, functions and parameters.
#
# Several features can be enabled using '#!define WITH_FEATURE' directives:
@@ -321,7 +321,7 @@ modparam("registrar", "max_expires", 3600)
modparam("acc", "early_media", 0)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 0)
-/* by default ww do not adjust the direct of the sequential requests.
+/* by default we do not adjust the direct of the sequential requests.
if you enable this parameter, be sure the enable "append_fromtag"
in "rr" module */
modparam("acc", "detect_direction", 0)
@@ -984,7 +984,7 @@ route[PSTN] {
#!ifdef WITH_PSTN
# check if PSTN GW IP is defined
if (strempty($sel(cfg_get.pstn.gw_ip))) {
- xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n");
+ xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n");
return;
}
@@ -1036,7 +1036,7 @@ route[TOVOICEMAIL] {
# check if VoiceMail server IP is defined
if (strempty($sel(cfg_get.voicemail.srv_ip))) {
- xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n");
+ xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n");
return;
}
if($avp(oexten)==$null)
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Module: kamailio
Branch: master
Commit: 28f01c863b2533b077be634f966dc43c113e2677
URL: https://github.com/kamailio/kamailio/commit/28f01c863b2533b077be634f966dc43…
Author: �������������������� �������������������������������� <git-dpa(a)aegee.org>
Committer: Daniel-Constantin Mierla <miconda(a)gmail.com>
Date: 2023-04-10T08:05:56+02:00
benchmark: typo
---
Modified: src/modules/benchmark/doc/benchmark_devel.xml
---
Diff: https://github.com/kamailio/kamailio/commit/…
[View More]28f01c863b2533b077be634f966dc43…
Patch: https://github.com/kamailio/kamailio/commit/28f01c863b2533b077be634f966dc43…
---
diff --git a/src/modules/benchmark/doc/benchmark_devel.xml b/src/modules/benchmark/doc/benchmark_devel.xml
index 4774e5fe46b..01a6aaf6b19 100644
--- a/src/modules/benchmark/doc/benchmark_devel.xml
+++ b/src/modules/benchmark/doc/benchmark_devel.xml
@@ -32,7 +32,7 @@
<function moreinfo="none">bm_register(name, mode, id)</function>
</title>
<para>
- This function register a new timer and/or returns the internal ID
+ This function registers a new timer and/or returns the internal ID
associated with the timer. mode controls the creation of new timer
if not found. id is to be used by start and log timer functions.
</para>
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Module: kamailio
Branch: master
Commit: b26cfeda773867e8f346f3eb0f5c31e8cd408ac9
URL: https://github.com/kamailio/kamailio/commit/b26cfeda773867e8f346f3eb0f5c31e…
Author: �������������������� �������������������������������� <git-dpa(a)aegee.org>
Committer: Daniel-Constantin Mierla <miconda(a)gmail.com>
Date: 2023-04-10T08:03:47+02:00
rtpengine clarifications
See http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/2023-April/0… and
https://github.com/sipwise/rtpengine/…
[View More]issues/1595#issuecomment-1382378886
---
Modified: src/modules/rtpengine/doc/rtpengine_admin.xml
---
Diff: https://github.com/kamailio/kamailio/commit/b26cfeda773867e8f346f3eb0f5c31e…
Patch: https://github.com/kamailio/kamailio/commit/b26cfeda773867e8f346f3eb0f5c31e…
---
diff --git a/src/modules/rtpengine/doc/rtpengine_admin.xml b/src/modules/rtpengine/doc/rtpengine_admin.xml
index ba6e9f2d22b..0cb881b4397 100644
--- a/src/modules/rtpengine/doc/rtpengine_admin.xml
+++ b/src/modules/rtpengine/doc/rtpengine_admin.xml
@@ -2366,7 +2366,8 @@ rtpengine_offer();
<emphasis>force-answer</emphasis> - force <quote>answer</quote>, that is,
only rewrite &sdp; when corresponding session already exists
in the &rtp; proxy. By default is on when the session is to be
- completed.
+ completed. This is only necessary when the offer was sent by <function>rtpengine_offer()</function>, and
+ the answer is handled by <function>rtpengine_manage()</function>.
</para></listitem>
<listitem><para>
<emphasis>direction=...</emphasis> - this option specifies a logical network
@@ -2482,9 +2483,9 @@ rtpengine_offer();
transport protocol that should be used towards the recipient of
the &sdp;. If none of them are specified, the protocol given in
the &sdp; is left untouched. Otherwise, the <quote>SRTP</quote> flag indicates that
- SRTP should be used, while <quote>RTP</quote> indicates that SRTP should not be used.
- <quote>AVPF</quote> indicates that the advanced RTCP profile with feedback messages
- should be used, and <quote>AVP</quote> indicates that the regular RTCP profile
+ SRTP should be used, while <quote>RTP</quote> indicates that both SRTP and AVPF should
+ not be used. <quote>AVPF</quote> indicates that the advanced RTCP profile with feedback
+ messages should be used, and <quote>AVP</quote> indicates that the regular RTCP profile
should be used. See also the next set of flags below.
</para></listitem>
<listitem><para>
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I have installed kamaillio 5.6 on debian 11, bullseye with RTP engine latest version 11.x. Using webrtc client as jssip to connect with kamailio and route the calls to asterisk server from kamailio.
in Local network everything working fine
When I connect JSSIP webrtc client from public internet I am getting no voice for both ends. I have observed that the rtp is flowing from the remote end to the client while IVR is playing from callee end. But the rtp doesnt know where to route to reach …
[View More]the webrtc client.
For normal UDP call everyting works fine from local and public internet. The issue is only for ws and wss calls
observations :
ICE options : trickle ICE
ICE candidates with sdp is there when calling from local network
ICE candidates are not with sdp when calling from publi internet. C IN and RTCP = 0..0.0.0:9
found RTP is routing to ------- > 0.0.0.0:9 and not reaching anywhere
RTPengine interface is local ip and public ip is advertised, in kmailio also public ip is advertised
sip dumb has been investgated and found everything is observed as normal with only one difference, that is no ice candidates for wss client from public internet
--
Reply to this email directly or view it on GitHub:
https://github.com/kamailio/kamailio/issues/3413
You are receiving this because you are subscribed to this thread.
Message ID: <kamailio/kamailio/issues/3413(a)github.com>
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Module: kamailio
Branch: master
Commit: a4970749b2be40cd598d4ce5cf2eaa6b38fc2307
URL: https://github.com/kamailio/kamailio/commit/a4970749b2be40cd598d4ce5cf2eaa6…
Author: Ovidiu Sas <osas(a)voipembedded.com>
Committer: Ovidiu Sas <osas(a)voipembedded.com>
Date: 2023-04-08T01:48:11-04:00
tm: enhance documentation for on_sl_reply param
---
Modified: src/modules/tm/doc/params.xml
---
Diff: https://github.com/kamailio/kamailio/commit/a4970749b2be40cd598d4ce5cf2eaa6…
Patch: https:…
[View More]//github.com/kamailio/kamailio/commit/a4970749b2be40cd598d4ce5cf2eaa6…
---
diff --git a/src/modules/tm/doc/params.xml b/src/modules/tm/doc/params.xml
index e19d709e5a8..5b65dcabebe 100644
--- a/src/modules/tm/doc/params.xml
+++ b/src/modules/tm/doc/params.xml
@@ -715,7 +715,7 @@ modparam("tm", "reparse_on_dns_failover", 0)
</para>
<para>
If KEMI is used, the parameter has to be the KEMI function name, which
- receives a string parameter with the event name.
+ receives a string parameter with the event name, respectively "on_sl_reply".
</para>
<example>
<title>Set <varname>on_sl_reply</varname> parameter</title>
@@ -727,6 +727,20 @@ modparam("tm", "on_sl_reply", "stateless_replies")
onreply_route["stateless_replies"] {
# do not allow stateless replies to be forwarded
return 0;
+}
+ </programlisting>
+ </example>
+ <example>
+ <title>Set <varname>on_sl_reply</varname> parameter (KEMI javascript)</title>
+ <programlisting>
+...
+modparam("tm", "on_sl_reply", "ksr_slreply")
+...
+-- event callback function implemented in javascript
+function ksr_slreply(evname) {
+ # the evname is set to "on_sl_reply"
+ # do not allow stateless replies to be forwarded
+ KSR.x.drop();
}
</programlisting>
</example>
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