Thu Nov 16 07:23:30 Eastern Standard Time 2006 Pekka.Pessi@nokia.com
* nta.c: setting the local sequence number of nta_leg_t only when first reques
t is sent.
Application can now set the initial value of CSeq either in nta_leg_create()
or in nta_outgoing_*create() (or nta_msg_request_complete()).
* nua_session.c: fixed session timer negotiation when UAS does refreshing with
INVITEs
The session-expires header had "uac" even when uac did not support timer.
The UAS failed to send re-INVITEs.
Thanks for Chung Pak Lai for reporting this problem.
* bnf: added host_cmp().
* outbound.c: using host_cmp() to check if Via host and received parameter dif
fer
Bug reported by Marc Blanchet.
* nua_session.c: fixed leak in incomin INVITE processing.
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3425 d0543943-73ff-0310-b7d9-9358b9ac24b2
This addition lets you set artifical ringback on a channel
that is waiting for an originated call to be answered.
the syntax is
<action application="set" data="ringback=[data]"/>
where data is either the full path to an audio file
or a teletone generation script..
syntax of teletone scripts
LEGEND:
0-9,a-d,*,# (standard dtmf tones)
variables: c,r,d,v,>,<,+,w,l,L,%
c (channels) - Sets the number of channels.
r (rate) - Sets the sample rate.
d (duration) - Sets the default tone duration.
v (volume) - Sets the default volume.
> (decrease vol) - factor to decrease volume by per frame (0 for even decrease across duration).
< (increase vol) - factor to increase volume by per frame (0 for even increase across duration).
+ (step) - factor to step by used by < and >.
w (wait) - default silence after each tone.
l (loops) - number of times to repeat each tone in the script.
L (LOOPS) - number of times to repeat the the whole script.
% (manual tone) - a generic tone specified by a duration, a wait and a list of frequencies.
standard tones can have custom duration per use with the () modifier
7(1000, 500) to generate DTMF 7 for 1 second then pause .5 seconds
EXAMPLES
UK Ring Tone [400+450 hz on for 400ms off for 200ms then 400+450 hz on for 400ms off for 2200ms]
%(400,200,400,450);%(400,2200,400,450)
US Ring Tone [440+480 hz on for 2000ms off for 4000ms]
%(2000,4000,440,480)
ATT BONG [volume level 4000, even decay, step by 2, # key for 60ms with no wait, volume level 2000, 350+440hz {us dialtone} for 940ms
v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440)
SIT Tone 913.8 hz for 274 ms with no wait, 1370.6 hz for 274 ms with no wait, 1776.7 hz for 380ms with no wait
%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)
ATTN TONE (phone's off the hook!) 1400+2060+2450+2600 hz for 100ms with 100ms wait
%(100,100,1400,2060,2450,2600)
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3408 d0543943-73ff-0310-b7d9-9358b9ac24b2
This is some brand new stuff to gateway chat/presence/audio from one protocol to another
So far it only works between google/jingle and SIP
All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end.
With this setup registered X-Lite's can chat with each other and call each other
as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls.
Chat May also be done between X-Lite and jabber
You'll also need a jabber server configured for component login so you can interface.
We have only tested with jabberd2 so far.
Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example)
so the jabber records are pointed at your jabber server.
RELEVANT CONFIGS
<!-- Brian has no jingle support so send calls to him over to his iax url -->
<extension name="bkw">
<condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$">
<action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/>
</condition>
</extension>
<!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below -->
<extension name="jingle2sip">
<condition field="source" expression="mod_dingaling"/>
<condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$">
<action application="bridge" data="sofia/$2/$1%$2"/>
</condition>
</extension>
<extension name="sip2jingle">
<condition field="source" expression="mod_sofia"/>
<condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$">
<action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/>
</condition>
</extension>
<configuration name="sofia.conf" description="sofia Endpoint">
<global_settings>
<param name="log-level" value="0"/>
</global_settings>
<profiles>
<profile name="fs.mydomain.com">
<registrations/>
<settings>
<param name="debug" value="1"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="PCMU"/>
<param name="codec-ms" value="20"/>
<param name="accept-blind-reg" value="true"/>
<param name="manage-presence" value="true"/>
<!--<param name="full-id-in-dialplan" value="true"/>-->
<!--<param name="auth-calls" value="true"/>-->
<!--<param name="auth-all-packets" value="true"/>-->
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="rtp-ip" value="100.200.100.200"/>
<param name="sip-ip" value="fs.mydomain.com"/>
</settings>
</profile>
</profiles>
</configuration>
<configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
<settings>
<param name="debug" value="0"/>
<param name="codec-prefs" value="PCMU"/>
</settings>
<profile type="component">
<param name="name" value="fs.mydomain.com"/>
<param name="password" value="secret"/>
<param name="dialplan" value="XML"/>
<param name="rtp-ip" value="208.64.200.42"/>
<param name="server" value="jabber.freeswitch.org:5347"/>
<!-- disable to trade async for more calls -->
<param name="use-rtp-timer" value="true"/>
<param name="exten" value="_auto_"/>
<!--<param name="vad" value="both"/>-->
</profile>
</configuration>
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
Michael Jerris