Hello Ser Users,
I recently installed SER with the mysql module and all worked well. I
could make calls to myself,etc
But when i tried to add on the ACC Module, it started well , i could
telnet it(telnet localhost 5060) but can't login in any more( i use
x-lite softphones from xten.com). The error is "Login failed Contact
network admin!" and when i try to call, the error is "Call failed: 500
I'm terribly sorry, server error occl". sometimes the logging in error
is "Login timed out". Those errors are from the softphone, but the SER
server gives no errors..even nothing is recorded in the system logs,
Note: I installed the ACC Module by addimg this to ser.cfg:
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
and to the routing block
if (uri=~"sip:+49") {
if (!proxy_authorize("iptel.org", "subscriber")) {
proxy_challenge("iptel.org", "0");
break;
};
if (method=="INVITE") {
log("from!=digest\n");
sl_send_reply("403","Forbidden");
break;
};
setflag(1);
t_relay();
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
Regards,
Wilson
---
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.282 / Virus Database: 150 - Release Date: 9/25/2001
Hi All,
I'm a SER newbie and want to be able to setup a SIP conversation for
office use. I'm using softphone from xten and already installed SER in my
linux box.
Question:
As a very eager user, I want to make SIP calls at once. I have two
workstations with Softphone (xten) clients. Can someone give me some
pointers on the minimum configuration to make calls between the two
workstations?
TIA.
Or you can also use CPL.
-----Original Message-----
From: Jev [mailto:jev@ecad.org]
Sent: Friday, August 27, 2004 6:24 PM
To: Vitaly Nikolaev
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] one to many aliases and voicemail
Hi Vitaly, thanks for your reply,
So, if I follow what you are saying;
after the call fails (when fvr_timer value elapses for instance) and the
call gets passed to the failure_route and then to voicemail, i will
appear as a call to alias@sipdomian.
That would mean that I could put
if (uri!=~"^sip:1111@.*") { #Don't use voicemail on 1111 alias
t_on_failure("1"); #Our voicemail failure route
}
But then, what about aliases that map to a single user, and that user
has voicemail. When sems gets the call to ALIAS to won't know what
account that alias is associated with unless it looks at the subscriber
table, and in which event it is possible that it will find more than one
account that will receive voicemail.
What do you think?
-Jev
Vitaly Nikolaev wrote:
> U have incoming DID number 1111 that forking to to aliases 2222 and
3333
>
> You have failure route configured on the ser and there u have
forwarding
> to voicemail
>
> Now.. if somebody call 1111 SER will fork call to 2222 and 3333 and if
> both of them did not pickup the phone it will be passed to failure
route
> BUT not as call to 2222/3333 but as call to 1111
>
> That means that you should not care about configure voicemail on your
> aliases.. only configure VM or not on incoming DID
>
>
> I hope it is understandable :)
>
>
>
>
>
>>-----Original Message-----
>>From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]
>
> On
>
>>Behalf Of Jev
>>Sent: Friday, August 27, 2004 3:30 PM
>>To: serusers(a)lists.iptel.org
>>Subject: [Serusers] one to many aliases and voicemail
>>
>>Hi all,
>>
>>Having an alias map to a single uri is fine, and if the target uri has
>>voicemail enabled that is fine.
>>
>>What happens if I have a alias that goes one to many, and some of the
>>target uris have voicemail enabled? When someone calls a particular
>>alias, I don't want ser to redirect to sems at all.
>>
>>How can I achieve this? Is it possible to detect if a invite has the
>>result of an alias (probably by using hints in the sip msg?), and if
>
> so
>
>>can I detect how many targets a alias maps to? In basic terms, if
>>alias_targets is > 1, then do not redirect to voicemail.
>>
>>Thanks!
>>-Jev
>>
>>_______________________________________________
>>Serusers mailing list
>>serusers(a)lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>
>
>
>
I would recommend you to use exec_dset procedure and to have external
script that will do all routing for you.
-----Original Message-----
From: Jev [mailto:jev@ecad.org]
Sent: Friday, August 27, 2004 6:24 PM
To: Vitaly Nikolaev
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] one to many aliases and voicemail
Hi Vitaly, thanks for your reply,
So, if I follow what you are saying;
after the call fails (when fvr_timer value elapses for instance) and the
call gets passed to the failure_route and then to voicemail, i will
appear as a call to alias@sipdomian.
That would mean that I could put
if (uri!=~"^sip:1111@.*") { #Don't use voicemail on 1111 alias
t_on_failure("1"); #Our voicemail failure route
}
But then, what about aliases that map to a single user, and that user
has voicemail. When sems gets the call to ALIAS to won't know what
account that alias is associated with unless it looks at the subscriber
table, and in which event it is possible that it will find more than one
account that will receive voicemail.
What do you think?
-Jev
Vitaly Nikolaev wrote:
> U have incoming DID number 1111 that forking to to aliases 2222 and
3333
>
> You have failure route configured on the ser and there u have
forwarding
> to voicemail
>
> Now.. if somebody call 1111 SER will fork call to 2222 and 3333 and if
> both of them did not pickup the phone it will be passed to failure
route
> BUT not as call to 2222/3333 but as call to 1111
>
> That means that you should not care about configure voicemail on your
> aliases.. only configure VM or not on incoming DID
>
>
> I hope it is understandable :)
>
>
>
>
>
>>-----Original Message-----
>>From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]
>
> On
>
>>Behalf Of Jev
>>Sent: Friday, August 27, 2004 3:30 PM
>>To: serusers(a)lists.iptel.org
>>Subject: [Serusers] one to many aliases and voicemail
>>
>>Hi all,
>>
>>Having an alias map to a single uri is fine, and if the target uri has
>>voicemail enabled that is fine.
>>
>>What happens if I have a alias that goes one to many, and some of the
>>target uris have voicemail enabled? When someone calls a particular
>>alias, I don't want ser to redirect to sems at all.
>>
>>How can I achieve this? Is it possible to detect if a invite has the
>>result of an alias (probably by using hints in the sip msg?), and if
>
> so
>
>>can I detect how many targets a alias maps to? In basic terms, if
>>alias_targets is > 1, then do not redirect to voicemail.
>>
>>Thanks!
>>-Jev
>>
>>_______________________________________________
>>Serusers mailing list
>>serusers(a)lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>
>
>
>
Hi all
I am using SER+NAT+rtpproxy as proxy servers. I have serched
seruser mailing list for solution and I didn't find any. Please help
me in below senerio.
Elements
-----------------
1) UA 'PUBLIC' ( in public internet with IP
211.95.122.XXX.XXX)
2) SIP SER with NAT helper(CVS version) + rtp proxy on same
machine(http://www.portaone.com/~sobomax/rtpproxy.tar ) with one
external public IP and internal privite IP)
3) UA 'PRIVATE' - in side private domain with 192.17.1.XXX) ip address.
4) UA 'PUBLIC' not register to ser. UA 'PRIVATE' have registerd to ser.
I am trying to accomplish to make call from UA 'PRIVATE' to UA 'PUBLIC'.
i'e
calls from private IP to public IP using SER as my signalling proxy
and rtpproxy as proxy for RTP.
SER is succussfully etablishing connection between two user agents and
but it is failed to forward rtp between two UAs. No voice. Request as
'INVITE'
ser proxy change SDP's ip and port, but reply as stats 200 and 183 did
not change
SDP's ip and report, so UA 'PUBLIC' can hear UA 'PRIVATE', but UA
'PRIVATE' can't
hear from UA 'PUBLIC'.
Is it MUST that UA 'PUBLIC' register to ser?
This is my ser.cfg:
# main routing logic
route
{
/*================================ ROUTINE CHECKS
=============================================*/
# filter too old messages
if (!mf_process_maxfwd_header("10"))
{
log("LOG: Too many hops\n");
sl_send_reply("483","Too Many Hops");
break;
}
if (msg:len > max_len)
{
sl_send_reply("513", "Message too big");
break;
}
# compulsory processing of Route header fields and adding RR
loose_route();
# These devices need rtp proxy
#if (search("User-Agent:"))
#{
setflag(2); # remember this needs rtp proxy
force_rport();
# fix_nated_contact();
#}
# registration (uses rewritten contacts)
if (method=="REGISTER")
{
if(!radius_proxy_authorize("REGISTER"))
{
sl_send_reply("401", "Forbidden");
break;
}
save("location");
break;
}
# Invite authentication
if (method=="INVITE")
{
record_route();
if (isflagset(2))
{
if(!radius_proxy_authorize("INVITE"))
{
sl_send_reply("401", "Forbidden");
break;
}
force_rtp_proxy();
}
/* set up reply processing */
t_on_reply("1");
}
# Forward
if(uri=~"^sip:5[0-9][0-9]*@")
{
rewritehost("211.95.122.158");
prefix("66123");
forward(211.95.122.158, 5060);
break;
}
# Look up the callee
if (method == "INVITE" || method == "CANCEL")
{
if (!lookup("location"))
{
sl_send_reply("404", "Not Found");
break;
}
}
# forward the request statefuly now; (we need *stateful*
forwarding,
# because the stateful mode correlates requests with replies and
# drops retranmissions; otherwise, we would have to report on
# every single message received)
setflag(1); # set for accounting (the same value as in log_flag!)
if(!t_relay())
{
sl_reply_error();
break;
}
}
# all incoming replies for t_onrepli-ed transactions enter here
onreply_route[1]
{
if (status=~"(183)|2[0-9][0-9]")
{
force_rtp_proxy();
}
}
Thanks for help.
Regards,
robert
Hi all
I am using SER+NAT+rtpproxy as proxy servers. I have serched
seruser mailing list for solution and I didn't find any. Please help
me in below senerio.
Elements
-----------------
1) UA 'PUBLIC' ( in public internet with IP
211.95.122.XXX.XXX)
2) SIP SER with NAT helper(CVS version) + rtp proxy on same
machine(http://www.portaone.com/~sobomax/rtpproxy.tar ) with one
external public IP and internal privite IP)
3) UA 'PRIVATE' - in side private domain with 192.17.1.XXX) ip address.
4) UA 'PUBLIC' not register to ser. UA 'PRIVATE' have registerd to ser.
I am trying to accomplish to make call from UA 'PRIVATE' to UA 'PUBLIC'.
i'e
calls from private IP to public IP using SER as my signalling proxy
and rtpproxy as proxy for RTP.
SER is succussfully etablishing connection between two user agents and
but it is failed to forward rtp between two UAs. No voice. Request as
'INVITE'
ser proxy change SDP's ip and port, but reply as stats 200 and 183 did
not change
SDP's ip and report, so UA 'PUBLIC' can hear UA 'PRIVATE', but UA
'PRIVATE' can't
hear from UA 'PUBLIC'.
Is it MUST that UA 'PUBLIC' register to ser?
This is my ser.cfg:
# main routing logic
route
{
/*================================ ROUTINE CHECKS
=============================================*/
# filter too old messages
if (!mf_process_maxfwd_header("10"))
{
log("LOG: Too many hops\n");
sl_send_reply("483","Too Many Hops");
break;
}
if (msg:len > max_len)
{
sl_send_reply("513", "Message too big");
break;
}
# compulsory processing of Route header fields and adding RR
loose_route();
# These devices need rtp proxy
#if (search("User-Agent:"))
#{
setflag(2); # remember this needs rtp proxy
force_rport();
# fix_nated_contact();
#}
# registration (uses rewritten contacts)
if (method=="REGISTER")
{
if(!radius_proxy_authorize("REGISTER"))
{
sl_send_reply("401", "Forbidden");
break;
}
save("location");
break;
}
# Invite authentication
if (method=="INVITE")
{
record_route();
if (isflagset(2))
{
if(!radius_proxy_authorize("INVITE"))
{
sl_send_reply("401", "Forbidden");
break;
}
force_rtp_proxy();
}
/* set up reply processing */
t_on_reply("1");
}
# Forward
if(uri=~"^sip:5[0-9][0-9]*@")
{
rewritehost("211.95.122.158");
prefix("66123");
forward(211.95.122.158, 5060);
break;
}
# Look up the callee
if (method == "INVITE" || method == "CANCEL")
{
if (!lookup("location"))
{
sl_send_reply("404", "Not Found");
break;
}
}
# forward the request statefuly now; (we need *stateful*
forwarding,
# because the stateful mode correlates requests with replies and
# drops retranmissions; otherwise, we would have to report on
# every single message received)
setflag(1); # set for accounting (the same value as in log_flag!)
if(!t_relay())
{
sl_reply_error();
break;
}
}
# all incoming replies for t_onrepli-ed transactions enter here
onreply_route[1]
{
if (status=~"(183)|2[0-9][0-9]")
{
force_rtp_proxy();
}
}
Thanks for help.
Regards,
robert
When i am using in the configuration file
modparam("usrloc", "db_mode", 1)
modparam("usrloc", "db_url","sql://root:hello@localhost/ser")
i get ERROR: receive_msg: no mem for sip_msg
i am using ser 0.8.14
when i use
modparam("usrloc", "db_mode", 0)
there is not any problem in the server
Could any one help???
Thanks
hello friends,
iam using sems-0.1.0 and ser-0.8.12 my main aim to place the announcement if at all any body call to the number who is not logged in so my ser.cfg looks like this
but in dialer ( xlite ) show s trying for fewminutes then says timed out.
ser.cfg
----------
debug=7 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=yes # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
debug=7
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
#port=5060
#children=4
fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tmloadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/vm.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
modparam("usrloc","db_url","mysql://ser:heslo@localhost:3306/ser".so"
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# ------------------------- request routing logic -------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
save("location");
break;
};
if ( method=="ACK" || method=="INVITE" || method=="BYE" || method=="CANCEL" )
{
if (!t_newtran())
{
sl_send_reply("500","could not create transaction");
break;
};
t_reply("100","Trying - just wait a minute !");
if (method=="INVITE")
{
log("in to the invite");
if(!vm("/tmp/am_fifo","announcement")
log("in to the invite");
if(!vm("/tmp/am_fifo","announcement"))
{
log("could not contact announcement server");
t_reply("500","error contacting sems");
};
break;
}
else if (method=="BYE" || method=="CANCEL")
{
if(!vm("/tmp/am_fifo","bye"))
{
t_reply("500","error contacting sems");
};
break;
};
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
}
-----------
and my sems.conf looks like this
-----------
# $Id: sems.conf.sample,v 1.1.2.1 2003/08/28 19:13:25 rco Exp $
#
# sems.conf.sample
#
# Sip Express Media Server (sems)
#
# sample configuration file
#
#
# whitespaces (spaces and tabs) are ignored
# comments start with a "#" and may be used inline
#
# example: option=value1, value2 # i like this option
#
##################################
# global parameters #
##################################
# optional parameter: fork={yes|no}
#
# - specifies if sems should run in daemon mode (background)
fork=yes
# optional parameter: stderr={yes|no}
#
# - debug mode: do not fork and log to stderr
stderr=no
# optional parameter: loglevel={0|1|2|3}
#
# - sets log level (error=0, warning=1, info=2, debug=3)
# - sets log level (error=0, warning=1, info=2, debug=3)
loglevel=3
# optional parameter: fifo_name=<filename>
#
# - path and file name of our fifo file
fifo_name=/tmp/am_fifo
# optional parameter: ser_fifo_name=<filename>
#
# - path and file name of Ser's fifo file
ser_fifo_name=/tmp/ser_fifo
# optional parameter: plugin_path=<path>
#
# - sets the path to the plug-ins
# - may be absolute or relative to CWD
plugin_path=/usr/local/lib/sems/plug-in/
##################################
# voicemail specific parameters #
##################################
# optional parameter: announce_path=<path>
#
# - sets the path where announce files are searched for
announce_path=/usr/local/lib/sems/audio/
# optional parameter: default_announce=<filename>
#
# - sets the name of the default announce WAV file
default_announce=default_en.wav
# - sets the name of the default announce WAV file
default_announce=default_en.wav
# optional parameter: max_record=<seconds>
#
# - maximum record time
max_record=30
# optional parameter: smtp_server=<hostname>
#
# - sets address of smtp server
smtp_server=localhost
# optional parameter: smtp_port=<port>
#
# - sets port of smtp server
smtp_port=25
##################################
# module specific parameters #
##################################
# add more module configurations here (inline or external):
#
# config.mymodule=<filename>
# or
# config.mymodule=inline
# ...
# config.mymodule=end
-----------
the logs are as follows
------
12(15776) vm: calculated route:
12(15776) vm: next r-uri: sip:216105@xxx.xxx.xxx.18:5060
12(15776) parse_headers: flags=-1
12(15776) query="select email_address from subscriber where username='214103'"
12(15776) vm: write_to_vm_fifo: <0.2
sip_request.announcement
INVITE
214103
varala(a)xxx.xxx.in
xxx.xxx.xxx.in
xxx.xxx.xxx.19
.
sip:214103@xxx.xxx.xxx.in
sip:216105@xxx.xxx.xxx.18:5060
hello <sip:216105@xxx.xxx.xxx.in>
<sip:214103@xxx.xxx.xxx.in>
D4DD8ACF-F8FE-11D8-8FE6-00E018FCDC33(a)xxx.xxx.xxx.18
3583624574
.
48216
14502:756140263
.
.
P-MsgFlags: 0
.
v=0
o=216105 15926530 15926613 IN IP4 xxx.xxx.xxx.18
s=X-Lite
c=IN IP4 xxx.xxx.xxx.18
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
>
12(15776) DEBUG: write_to_vm_fifo: write completed
12(15776) DEBUG: add_to_tail_of_timer[0]: 0x422b9304
12(15776) receive_msg: cleaning up
14(15786) udp_rcv_loop: probing packet received from xxx.xxx.xxx.18 50195
15(15790) udp_rcv_loop: probing packet received from xxx.xxx.xxx.18 50195
18(15802) DEBUG: timer routine:0,tl=0x422b9304 next=(nil)
18(15802) DEBUG: FR_handler:stop retr. and send CANCEL (0x422b91c8)
18(15802) ->>>>>>>>> T_code=100, new_code=408
18(15802) DEBUG: relay_reply: branch=0, save=0, relay=0
18(15802) parse_headers: flags=-1
18(15802) check_via_address(xxx.xxx.xxx.18, xxx.xxx.xxx.18, 0)
18(15802) DEBUG: reply relayed. buf=0x80c3d38: SIP/2.0 4..., shmem=0x422ba890: SIP/2.0 4
18(15802) DEBUG: add_to_tail_of_timer[4]: 0x422b927c
18(15802) DEBUG: add_to_tail_of_timer[0]: 0x422b9290
18(15802) DEBUG: final_response_handler : done
9(15761) SIP Request:
9(15761) method: <ACK>
9(15761) uri: <sip:214103@xxx.xxx.xxx.in>
9(15761) version: <SIP/2.0>
9(15761) parse_headers: flags=1
9(15761) Found param type 235, <rport> = <n/a>; state=6
9(15761) Found param type 232, <branch> = <z9hG4bKD4DD8AD0F8FE11D88FE600E018FCDC33>; state=16
9(15761) end of header reached, state=5
9(15761) parse_headers: Via found, flags=1
9(15761) parse_headers: this is the first via
9(15761) After parse_msg...
9(15761) preparing to run routing scripts...
9(15761) DEBUG : sl_filter_ACK: to late to be a local ACK!
9(15761) DEBUG : is_maxfwd_present: searching for max_forwards header
9(15761) parse_headers: flags=128
9(15761) DEBUG: add_param: tag=a6a1c5f60faecf035a1ae5b6e96e979a-9573
9(15761) end of header reached, state=29
9(15761) DEBUG: get_hdr_field: <To> [74]; uri=[sip:214103@xxx.xxx.xxx.in]
9(15761) DEBUG: to body [<sip:214103@xxx.xxx.xxx.in>]
9(15761) get_hdr_field: cseq <CSeq>: <48216> <ACK>
9(15761) DEBUG: is_maxfwd_present: value = 70
9(15761) DEBUG: add_param: tag=3583624574
9(15761) end of header reached, state=29
9(15761) parse_headers: flags=256
9(15761) DEBUG: get_hdr_body : content_length=0
9(15761) found end of header
9(15761) find_first_route(): No Route headers found
9(15761) loose_route(): There is no Route HF
9(15761) check_self - checking if host==us: 17==9 && [xxx.xxx.xxx.in] == [127.0.0.1]
9(15761) check_self - checking if port 5060 matches port 5060
9(15761) check_self - checking if host==us: 17==13 && [xxx.xxx.xxx.in] == [xxx.xxx.xxx.19]
9(15761) check_self - checking if port 5060 matches port 5060
9(15761) DEBUG: t_addifnew: msg id=2 , global msg id=0 , T on entrance=0xffffffff
9(15761) parse_headers: flags=-1
9(15761) parse_headers: flags=60
9(15761) t_lookup_request: start searching: hash=14502, isACK=1
9(15761) DEBUG: RFC3261 transaction matched, tid=D4DD8AD0F8FE11D88FE600E018FCDC33
9(15761) DEBUG: t_lookup_request: transaction found (T=0x422b91c8)
9(15761) DEBUG: cleanup_uacs: RETR/FR timers reset
9(15761) DEBUG: add_to_tail_of_timer[2]: 0x422b9210
9(15761) receive_msg: cleaning up
18(15802) DEBUG: timer routine:0,tl=0x422b9290 next=(nil)
18(15802) DEBUG: timer routine:4,tl=0x422b927c next=(nil)
18(15802) DEBUG: timer routine:2,tl=0x422b9210 next=(nil)
18(15802) DEBUG: wait_handler : removing 0x422b91c8 from table
18(15802) DEBUG: delete transaction 0x422b91c8
18(15802) DEBUG: wait_handler : done
--------
but i could not able to listen any audio , audi file is there in the correct path
with regards
ser die hard
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