Attached (includes test case modifications).
Thanks
--
M. Ranganathan
From ae8c5ab2cb89a8709c4f4a2b6ec0335cb4712822 Mon Sep 17 00:00:00 2001
From: M. Ranganathan <[EMAIL PROTECTED]>
Date: Wed, 20 Aug 2008 11:31:36 -0400
Subject: [PATCH] Fixes the following problems:
1. XML RPC port for sipxbridge (default) is generated.
2. RTP port range for sipxbridge is redundant (not used anymore) this field
belongs to sipxrelay.
3. Adds PCMA option for allowed codecs.
---
sipXconfig/neoconf/etc/sipxbridge/bridge-sbc.xml | 25 ++++----------------
.../admin/dialplan/sbc/bridge/sipxbridge.xml | 2 +-
2 files changed, 6 insertions(+), 21 deletions(-)
diff --git a/sipXconfig/neoconf/etc/sipxbridge/bridge-sbc.xml b/sipXconfig/neoconf/etc/sipxbridge/bridge-sbc.xml
index 554555c..427be86 100644
--- a/sipXconfig/neoconf/etc/sipxbridge/bridge-sbc.xml
+++ b/sipXconfig/neoconf/etc/sipxbridge/bridge-sbc.xml
@@ -98,6 +98,7 @@
<type>
<integer />
</type>
+ <value>8088</value>
<description>
Port for external XML/RPC requests. If not specified, the XML RPC server is not started. The XML RPC interface
is used by sipXconfig to report status to the user.
@@ -133,25 +134,6 @@
required by sipXecs only for calls that go through sipXbridge.
</description>
</setting>
- <setting name="rtp-port-range-start" advanced="yes">
- <label>Start RTP port</label>
- <profileName>$ignore$</profileName>
- <type>
- <integer required="yes" />
- </type>
- <value>25000</value>
- <description>Start of the RTP ports range.</description>
- </setting>
- <setting name="rtp-port-range-end" advanced="yes">
- <label>End RTP port</label>
- <profileName>$ignore$</profileName>
- <type>
- <integer required="yes" />
- </type>
- <value>25500</value>
- <description>End of the RTP ports range.</description>
- </setting>
- <setting name="rtp-port-range" hidden="yes"></setting>
<setting name="is-reinvite-supported" advanced="yes">
<label>re-INVITE support</label>
<type refid="switch" />
@@ -171,6 +153,9 @@
<value>PCMU</value>
</option>
<option>
+ <value>PCMA</value>
+ </option>
+ <option>
<value>G729</value>
</option>
<option>
@@ -180,7 +165,7 @@
</type>
<value>PCMU</value>
<description>
- The codec for the SDP offer that is allowed. This parameter is ONLY relevant if the ITSPs do NOT support
+ The codec for the SDP offer / answer that is allowed. This parameter is ONLY relevant if the ITSPs do NOT support
re-INVITE. Otherwise it is ignored. If the re-INVITE-supported flag is set to "false" then SDP offers and
answers are filtered to allow only this codec through.
</description>
diff --git a/sipXconfig/neoconf/test/org/sipfoundry/sipxconfig/admin/dialplan/sbc/bridge/sipxbridge.xml b/sipXconfig/neoconf/test/org/sipfoundry/sipxconfig/admin/dialplan/sbc/bridge/sipxbridge.xml
index 8664de7..0e83c7b 100644
--- a/sipXconfig/neoconf/test/org/sipfoundry/sipxconfig/admin/dialplan/sbc/bridge/sipxbridge.xml
+++ b/sipXconfig/neoconf/test/org/sipfoundry/sipxconfig/admin/dialplan/sbc/bridge/sipxbridge.xml
@@ -12,8 +12,8 @@
<stun-server-address>stun01.sipphone.com</stun-server-address>
<sip-keepalive-seconds>20</sip-keepalive-seconds>
<media-keepalive-seconds>1</media-keepalive-seconds>
+ <xml-rpc-port>8088</xml-rpc-port>
<music-on-hold-support-enabled>false</music-on-hold-support-enabled>
- <rtp-port-range>25000:25500</rtp-port-range>
<is-reinvite-supported>true</is-reinvite-supported>
<allowed-codec-name>PCMU</allowed-codec-name>
<route-inbound-calls-to-extension>operator</route-inbound-calls-to-extension>
--
1.5.4.1
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