Hi,all. I solved my problem. I added <action application="set" data="ringback=${us-ring}"/> into teh extension where i send call to sipx. Now i hear rbt for transit h323 - sip calls. Rgds, Nikolay. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Nikolay Kondratyev Sent: Tuesday, August 31, 2010 6:50 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] no ringback tone for transit h323 - sip call Hi all, I'm rather new to freeswitch, and i'm trying to setup FS (version 1.0.6) as sip-h323 gateway between sipX and avaya ip406. I have teh following test setup: sipX 172.23.9.2 beaver.sip.nstel.ru with extension 2853 (linksys spa 942 phone) Freeswitch 172.23.22.49 fs.lab.nstel.ru with extension 2854 (xlite) avaya ip406 with extension 5840 When i call 2853 -> 5840 i can hear ringback tone and voice is ok. When calling 5840 -> 2854, rbt and voice is ok. When calling 2854 -> 2853, rbt and voice is ok. When calling 5840 -> 2853, 2853 phone rings, there is no rbt, but voice is ok after answer. I have the following h323.conf file: [freeswitch@freeswitch autoload_configs]$ 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@beaver.sip.nstel.ru:5080"/ <mailto:sofia/external/$1@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.