[Freeswitch-users] no ringback tone for transit h323 - sip call

[Nikolay Kondratyev]

Hi,all.
I solved my problem.
I added 
<action application="set" data="ringback=${us-ring}"/>
into teh extension where i send call to sipx.
Now i hear rbt for transit h323 - sip calls.
Rgds,
Nikolay. 
  _____  

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nikolay Kondratyev
Sent: Tuesday, August 31, 2010 6:50 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] no ringback tone for transit h323 - sip call



Hi all,
I'm rather new to freeswitch, and i'm trying to setup FS (version 1.0.6) as sip-h323 gateway between sipX and avaya ip406.
 
I have teh following test setup:
sipX 172.23.9.2 beaver.sip.nstel.ru with extension 2853 (linksys spa 942 phone)
Freeswitch 172.23.22.49 fs.lab.nstel.ru with extension 2854 (xlite)
avaya ip406 with extension 5840
 
When i call 2853 -> 5840 i can hear ringback tone and voice is ok.
When calling 5840 -> 2854, rbt and voice is ok.
When calling 2854 -> 2853, rbt and voice is ok.
When calling 5840 -> 2853, 2853 phone rings, there is no rbt, but voice is ok after answer.
 
I have the following h323.conf file: 
[freeswitch at freeswitch autoload_configs]$ cat h323.conf.xml
<configuration name="h323.conf" description="Opal(h323) Endpoints">
  <settings>
    <param name="trace-level" value="4"/>
    <param name="context" value="default"/>
    <param name="dialplan" value="XML"/>
    <param name="codec-prefs" value="PCMA"/>
    <param name="gk-address" value=""/>    <!-- empty to disable, "*" to search LAN -->
    <param name="gk-identifer" value=""/>  <!-- optional name of gk -->
    <param name="gk-interface" value=""/>  <!-- optional listener interface name -->
  </settings>
  <listeners>
    <listener name="h323-listener">
      <param name="h323-ip" value="$${local_ip_v4}"/>
      <param name="h323-port" value="1720"/>
    </listener>
  </listeners>
</configuration>

in the default.xml dialplan i have default "Local_Extension" (with the only addition of a single number 2854 to be matched).
below Local_Extension i send 28xx numbers to sipx as following:
  <extension name="beaver">
    <condition field="destination_number" expression="^(28[0-9]{2})$">
      <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
      <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
      <action application="bridge" data="sofia/external/$1 at beaver.sip.nstel.ru:5080"/ <mailto:sofia/external/$1 at beaver.sip.nstel.ru:5080> >
    </condition>
  </extension>

FS logs for 5840 -> 2854 (rbt and voice ok) and 5840 -> 2853 (no rbt, voice ok) are attached.
In the log files i can see the moment when FS starts playing RBT into h323 channel in case 5840 -> 2854 call.
And i see that FS does not do it in case of 5840 -> 2853 call.
But i can not see why...
 
Probably i misconfigured something... 
Can anybody please help with this problem?
 
Thanks in advance,
Nikolay.
 
 
 

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