Hello,
I'm having a problem that is the following:
I'm using rewriteport to send traffic to another proxy. Then When I make a
call and if I hung up the other phone still rings.
What could be the problem?
Hi,
Here's a set of basic questions for which I need your help. I want to play
different announcements based on the caller's history. For simplicity, I
want ann1_user1.wav to be played for the first call, and ann2_user1.wav for
all subsequent calls from the 1st user. Similarly for 2nd user, the files
would be ann*_user2.wav.
1) I want to have a basic record of the number of times a user has called
the SER(ie, sent an INVITE). Which table and column in mySQL can I use for
this purpose. Also, how can I write the integer using SER commands (ie
something that I can put in ser.cfg file). There is an example (
http://www.iptel.org/write_custom_text_to_mysql_acc_table) which uses
'acc_db_request', but it can only write to sip_status column in acc table.
How can I modify /write to to any general field in mysql?
2) Subsequently, how can I retreive this value from mysql through some
command in ser.cfg file. I want to use this value in an 'if' statement and
redirect the process flow accordingly.
3) I was thinking of rewriting the uri to make it 'ann1_user1@xxxxx' and
sending the request to early_announce application so that the user hears
different announcement based on username and their call history (ie whether
this user has called before or not). However, I am not very clear about how
to do this. Is there any simpler method to do this. The final aim of this
whole exercise is to demonstrate call differentiation based on username and
user history. So if someone has a better idea then please do share it.
Thanks for the help and accept my apologies for squeezing so many questions
into a single message.
Regards,
Frq
Hi!
I have a problem with wesip, it works but it destroys client registration.
I added simple if statement, which executes
app on wesip server (only on INVITE):
#app server conn
if(!as_relay_t("examplesvr")){
t_reply("500","No App Server");
}
If you look at wesip log:
http://files.myopera.com/kriko/code/seas_log.txt
You can see that wesip doesn't process this request, since the app only
responds to INVITE, looking closer at this line:
---
SIP/2.0 481 Call Leg/Transaction Does Not Exist
---
If you would like to look at app it is here (really simple thing):
http://files.myopera.com/kriko/code/sample.java
What am I doing wrong? Is it because I'm using t_reply in wrong place or
there is something else. I don't think it is my app, since it is not being
executed.
--
(\__/)
(='.'=)
(")_(")
Hello Friends,
When Openser want to connect with disc client at that time disc client
close the connection and gives the folowing error.
DEBUG:peer_state_machine: peer 0x2aaaaaaf33a8, state = 3, event=0
ERROR:peer_state_machine: event - state mismatch : peer=0x2aaaaaaf33a8,
state=3, event=0
INFO:do_receive: timeout received
Actually Openser and DISC can not ESTABLISH the connection and gives
CLOSEWAIT in the netstat command.
So what will be problem ,please guide me.
Regards,
Dilip
Hi,
I am trying to understand the 'exec_msg' command as described in Section 4
of SER Admin Guide. I am just trying to echo the user name (1001) of the
caller on the screen. I use the config file given at the end. On receiving
the INVITE message, SER DOES echo the user name, but I also get some warning
and error messages as shown below:
#> /opt/ser-sems/sbin/ser -f /xxx.../voip/ser_expt.cfg
Listening on
udp: 192.168.3.208 [192.168.3.208]:5070
tcp: 192.168.3.208 [192.168.3.208]:5070
Aliases:
tcp: xxx:5070
udp: xxx:5070
WARNING: no fork mode
stateless - initializing
0(0) Maxfwd module- initializing
textops - initializing
exec - initializing
0(0) INFO: udp_init: SO_RCVBUF is initially 107520
0(0) INFO: udp_init: SO_RCVBUF is finally 262142
2(0) INFO: fifo process starting: 2803
2(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo...
2(0) WARNING: no fifo_db_url given - fifo DB commands disabled!
0(2801) Main Route...
0(2801) Inside search from-1001...
1001
0(2801) child process 2805 exited normally, status=0
0(2801) INFO: dont_fork turned on, living on
0(2801) WARNING: SIGPIPE received and ignored
0(2801) ERROR: exec_msg: cmd echo '1001' failed. exit_status=-1, errno=10:
No child processes
0(2801) WARNING: receive_msg: error while trying script
Can someone tell me how I can get rid of these messages. My aim was just to
get the user name displayed, and this has been achieved. I just want to
understand the reason for these warnings and error messages.
Thanks
Frq
PS: This is my config file:
# $Id: hello-world.cfg 46 2006-01-25 04:21:30Z /CN=Paul
Hazlett/emailAddress=paul(a)onsip.org $
debug=3
fork=no
log_stderror=yes
listen=192.168.3.208 #2.13 # put your server IP address here
port=5070
children=4
dns=no
rev_dns=no
fifo="/tmp/ser_fifo"
unix_sock="/tmp/ser_sock"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
#loadmodule "/opt/ser-sems/lib/ser/modules/mysql.so"
loadmodule "/opt/ser-sems/lib/ser/modules/sl.so"
loadmodule "/opt/ser-sems/lib/ser/modules/tm.so"
loadmodule "/opt/ser-sems/lib/ser/modules/rr.so"
loadmodule "/opt/ser-sems/lib/ser/modules/maxfwd.so"
loadmodule "/opt/ser-sems/lib/ser/modules/usrloc.so"
loadmodule "/opt/ser-sems/lib/ser/modules/registrar.so"
loadmodule "/opt/ser-sems/lib/ser/modules/textops.so"
loadmodule "/opt/ser-sems/lib/ser/modules/exec.so"
route {
log(1, "Main Route...\n");
# sl_send_reply("404", "User Not Found...:-(");
# if (uri=~"^sip:200@") {
# exec_msg("echo 'Frq'");
# }
if (search("(f|From): .*1002@")) {
log(3, "Inside search from-1002...\n");
exec_msg("echo '1002'");
};
if (search("(f|From): .*1000@")) {
log(3, "Inside search from-1000...\n");
exec_msg("echo '1000'");
};
if (search("(f|From): .*1001@")) {
log(3, "Inside search from-1001...\n");
exec_msg("echo 1001");
};
}
For those interested, you can monitor the progress of development for
the multi party chat functionality based on MSRP here:
http://www.msrprelay.org/http://www.msrprelay.org/wiki/MSRPChatSoftwareDesign
Adrian
>>>>>>>>>>>>
I've tryed Bria but in its web page it appears:
IM Management
- Send instant messages to groups
- Use IM tabs for multiple IM conversations in one window
- See IM alerts for incoming IM
It's not as a IM conferencing in a server side, but maybe enought...
Anyway, you can do a conference with IMC module and create contacts
like "IM
room 1", "IM room 2" in each client.
Of course, for now there is no way to know which users are joining a
conference. There are two RFC's about it:
- SIP Event Package for Conference State
http://tools.ietf.org/html/rfc4575
- SIP Call Control - Conferencing for User Agents
http://tools.ietf.org/html/rfc4579
but AFAIK there are not yet client side implementation.
--
Iñaki Baz Castillo
Hi
I'm new to openser and also Linux server install and configuration,
I'm profissional at windows and its network,
but need to install server for openser on local small lan.
also will be used as backup for data.
what is the easiest server to install?
--
hamed suhli
http://sourceforge.net/projects/tarmeez
This is a little off-topic, but figured it was a good place to ask.
We're interested in deploying OpenSER coupled with a SIP client
primarily used for IM, but also for voice and video conferencing. I'm
looking for feedback and recommendations on clients from people that may
do this already or have been involved with it.
So far one of the better ones looks like Bria (a step up from Eyebeam)
from CounterPath. One short coming though is the lack of IM
conferencing. I could probably rig something up with IMC but our users
would prefer to have a clicky GUI for it. This is for an enterprise and
it'd be nice to be able to brand it.
Thanks
-Bob
Hi folks,
When I am doing forking I am getting a puzzling result concerning
forwarding of responses, and I'm wondering how script needs tweaking.
When recieving a 6xx on one branch, the TM-module automatically
generates CANCEL to cancel the remaining branches(I wouldnt mind
if this was configurable), but the puzzling thing is that it
forwards the 603 _before_ it has terminated the remaining client
transactions. (It sends cancel but it doesnt wait for final responses
of the invites its cancelling, just forwards 6xx directly.)
A similar scenario happends when I let Timer C pop. My TM-module
will send a final response on the server-side at the same time as
it starts to cancel the client side.
But since one of the client txns just might result in a 2xx(INV)
I cant forward the final response until ALL client txns are done,
if I am going to follow the rfc properly.
I thought the TM module would keep track of all client txns for "me",
but it seems it isnt, so is this something I should "manually" handle
in my script?
Regards
Taisto Qvist
Hi all,
In find some informations about an automatic end of call system.
The aim is to cut the call after 2 hours. So it's not specificly link to
a pre-paid system, but for all my ser user, I want to end the call after
2 hours.
Is there a existing solution to do that ? Or maybe some idea to deal
with this project ?
Thank you !