Just to warn you we do not use mod_iax all that much. It was used as a tool to test the core of FreeSWITCH in the early stages of development and we just kinda left it hanging around since it does at least work. We had to add a bunch of code to the stock iaxclient lib to make it threadsafe etc because the lib is designed to handle 1 or 2 calls and we are using it in a b2bua type env where hundreds of concurrent calls may be possible. That said, since we more or less forked the code to avoid generating support issues for the other soft-phones that use iaxclient, you may want to check if there were any bugs in the real version of the library that match your issue.

http://sourceforge.net/projects/iaxclient

We've had little to no demand for IAX2 so we have not focused very much on it. We do have strategies for creating a more robust and perhaps even an "improved" implementation of the IAX2 protocol. Alas, it's pretty much purely an academic exercise because there is little reason to implement it.

Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400

----- Original Message ----
From: Birgit Arkesteijn <birgit@westhawk.co.uk>
To: Freeswitch-users@lists.freeswitch.org
Sent: Monday, March 10, 2008 10:52:40 AM
Subject: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable


Hi,

We're 
running 
FreeSWITCH 
svn 
Trunk 
7697 
on 
SuSE 
Linux 
9.3 
(x86-64).

I'm 
experiencing 
problems 
with 
DTMF:
The 
IAX2 
DTMF 
frames 
aren't 
always 
transferred 
into 
RTP 
packets.
Sometimes 
none 
of 
the 
frames 
are 
translated, 
sometimes 
only 
the 
first 
one 
is 
omitted. 
All 
the 
IAX2 
DTMF 
packets 
are 
ack-ed.

Our 
setup:
PhoneFromHere.com 
applet 
(IAX2 
protocol)
-> 
FreeSWITCH 
(ns2.westhawk.co.uk)
-> 
SIP 
to 
SystemX 
(sofia/default/123@66.212.134.192)


I 
searched 
on 
jira.freeswitch.org 
for 
a 
bug 
that 
could 
explain 
this, 
but 
failed, 
so 
I 
submitted 
this 
as 
a 
bug, 
see
http://jira.freeswitch.org/browse/FSCORE-109

Has 
anyone 
else 
experienced 
the 
same 
or 
a 
similar 
problem?

Thanks, 
Birgit

-- 
-- 
Birgit 
Arkesteijn, 
birgit@westhawk.co.uk,
-- 
Westhawk 
Ltd, 
Albion 
Wharf, 
19 
Albion 
Street, 
Manchester 
M1 
5LN, 
UK
-- 
Company 
no: 
1769350
-- 
Registered 
Office:
-- 
15 
London 
Road, 
Stockton 
Heath, 
Warrington 
WA4 
6SJ. 
UK.
-- 
tel.: 
+44 
(0)161 
237 
0660
-- 
<URL: 
http://www.westhawk.co.uk>

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