Le 15/02/2012 20:29, Michael Collins a écrit : Hello Michael,
show codec shows the codecs that FreeSWITCH can handle, but it does *not* show specific codecs your SIP profile is using. Look in your SIP profile (internal.xml or external.xml or whatever) and check out the codec_prefs.
I've fully reinstalled fs and sample configuration files (rm -rf $prefix, gmake install, gmake samples, gmake sounds/moh) The following files have been added to the base config. gateway definition file in $prefix/conf/sip_profiles/external :
<include> <gateway name="keyyo.net"> <param name="username" value="myname"/> <param name="password" value="mypw"/> <param name="realm" value="keyyo.net"/> <param name="proxy" value="keyyo.net"/> <param name="register-proxy" value="keyyo.net"/> <param name="register" value="true"/> <param name="ping" value="5"/> <param name="ping-max" value="10"/> <param name="ping-min" value="3"/> <param name="retry-seconds" value="5"/> </gateway> </include>
user definition file in $prefix/conf/directory/default
<include> <user id="1501"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1501"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1501"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="XLite Test"/> <variable name="effective_caller_id_number" value="1501"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="test"/> </variables> </user> </include>
extension for outgoing calls in £prefix/conf/diaplan/default
<include> <extension name="outbound_calls" continue="false"> <condition field="destination_number" expression = "^0(.*)$"> <action application="set" data="effective_caller_id_name=test"/> <action application="set" data="effective_caller_id_number=0980406292"/> <action application="bridge" data="sofia/gateway/keyyo.net/0$1"/> </condition> </extension> </include>
Except these three files, everything's out of the box setup. When dialing any number, the same codec negotiation issue pops up :
2012-02-16 19:50:40.735750 [DEBUG] sofia.c:5525 Channel sofia/external/0251685292 entering state [received][100] 2012-02-16 19:50:40.735750 [DEBUG] sofia.c:5536 Remote SDP: v=0^M o=root 14865 14867 IN IP4 192.168.22.103^M s=session^M c=IN IP4 83.136.164.34^M t=0 0^M m=audio 58298 RTP/AVP 8 101^M a=rtpmap:8 PCMA/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=ptime:20^M a=nortpproxy:yes^M
2012-02-16 19:50:40.735750 [DEBUG] sofia_glue.c:4820 Audio Codec Compare [PCMA:8:8000:20:64000]/[GSM:3:8000:20:13200] 2012-02-16 19:50:40.735750 [DEBUG] sofia_glue.c:4820 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] 2012-02-16 19:50:40.735750 [DEBUG] sofia_glue.c:4934 Set 2833 dtmf send payload to 101 2012-02-16 19:50:40.735750 [ERR] sofia.c:6036 Reinvite Codec Error! 2012-02-16 19:50:40.735750 [DEBUG] sofia.c:5525 Channel sofia/internal/1501@srvbsdfenssv.interne.associated-bears.org entering state [ready][200] 2012-02-16 19:50:40.735750 [DEBUG] switch_core_session.c:875 Send signal sofia/external/0251685292 [BREAK] 2012-02-16 19:50:40.735750 [DEBUG] switch_core_session.c:875 Send signal sofia/external/0251685292 [BREAK] 2012-02-16 19:50:40.735750 [DEBUG] switch_core_session.c:791 Send signal sofia/external/0251685292 [BREAK] 2012-02-16 19:50:40.735750 [DEBUG] switch_core_session.c:791 Send signal sofia/internal/1501@srvbsdfenssv.interne.associated-bears.org [BREAK]
So, any help solving this issue would be gladly received. Éric Masson