Hi,
I have installed ser-0.10.99-dev35-pa-4.1 and following up presence
handbook. i have configured XCAP using apache.
i am getting following errors with ser.cfg of ser package.
For PUBLISH request :
0(13411) get_pres_uri(): Error while extracting plain URI
0(13411) handle_publish(): Error while extracting presentity URI
For SUBSCRIBE request:
0(13411) ERROR: rls_handler.c:484: XCAP query problems
0(13411) XCAP query failed
0(13411) ERROR: rls_handler.c:484: XCAP query problems
0(13411) XCAP query failed
have any one succeed with presence? help me...
*-Venkat
*
Hi:
I use module AVPops and Textops do chang username of request uri.
example:
john(a)10.10.16.23
I hope chang => 4762(a)10.10.10.22
my configuration sample:
if (uri=~"john@")
{
avp_write("4762", "s:phone");
subst_user(/(.*)/$avp(phone)/');
rewritehostport("10.10.10.22:5060");
forward(uri:host, uri:port);
avp_delete("s:phone");
};
but reply is "Call failed:Device not found"
I try this sample is OK
if (uri=~"john@")
{
avp_write("4762", "s:phone");
subst_user(/(.*)/4762/');
rewritehostport("10.10.10.22:5060");
forward(uri:host, uri:port);
avp_delete("s:phone");
};
OS: Freebsd 5.4
SIP server: SER-0.9.6
please tell my problem is on? Thank you
Chungyu
This is my Configuration:
# ----------- global configuration parameters ------------------------
#debug=3 # debug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
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"
fifo_db_url="mysql://ser:heslo@localhost/ser"
# ------------------ 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/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/textops.so"
loadmodule "/usr/local/lib/ser/modules/avp.so"
#loadmodule "/usr/local/lib/ser/modules/uri.so"
#loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/avpops.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")
# -- 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 >= 2048 ) {
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
if (!method=="REGISTER") record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
break;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
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") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("xx.xx.xx.xx", "subscriber")) {
www_challenge("xx.xx.xx.xx", "0");
break;
};
save("location");
break;
};
if (uri=~"john@")
{
avp_write("4762","phone");
subst_user('/(.*)/$avp(phone)/');
rewritehostport("10.10.10.22:5060");
forward(uri:host, uri:port);
avp_delete("phone");
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
Can't make outbound calls, though not related directly to OpenSER..
Maybe someone here knows how to configure these guys. I have inbound
calls being automatically dialed into a * extension but If I make any
outbound calls I get a syslog output:
16:45:21.609 : 192.168.1.211 : WARNING : ( lgr_call)(458 ) !!
[ERROR] Call::GetEndPoint- Can't find endpoint for phone number
xxxxxx2018
16:45:21.593 : 192.168.1.211 : WARNING : ( lgr_psbrdif)(457 ) !!
[ERROR] AcBoard::GetEndPoint- Can't find EndPoint for Dest:xxxxxxx2018
Source:xxxxxxx11526 SourceIp:c0a80188
16:45:21.593 : 192.168.1.211 : WARNING : ( lgr_TrnkGrp)(456 ) !!
[ERROR] #0:TrunkGroup::AllocateEndPoint- Can't find EndPoint for phone
number xxxxxx2018
16:45:21.593 : 192.168.1.211 : NOTICE : ( lgr_stk_ses)(455 )
SIPStackSession::HandleStackSetupEV - NEWCALL: SrcPN=0
Any help would be great.
Also, caller ID doesn't work - only shows the caller ID of whatever is
set at the endpoint phone number settings and not what is actually
coming on the line :(
Nick
Hi,
I just compiled openSER with TLS support. I checked that TLS = 1 in the
Makefile when I compiled openSER. Now when I try to uncomment the parameters
in the openser.cfg to enable the TLS support and restart openSER it does not
start (I am using openserctl start command to start openser). It gives an
error saying ERROR:PID file /var/run/openser.pid does not exist -- OpenSER
start failed. I am using the following parameters in the openser.cfg file
for the TLS support:
disable_tls = 0
listen = tls:10.30.100.41:5061
tls_verify = 1
tls_require_certificate = 0
tls_method = TLSv1
tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem"
tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem"
tls_ca_list = "usr/local/etc/openser/tls/user/user-calist.pem"
I have checked that all the paths are correct in defining the
tls_certificate, tls_private_key and tls_ca_list.
I used the source tarball
openser-1.1.0-tls_src.tar.gz<http://www.openser.org/pub/openser/latest/src/openser-1.1.0-tls_src.tar.gz>
for
installing the openser. Your help is much appreciated.
Thanks
NCheeku
Hi all,
I am a SER/openSER newbie and was looking at an option of using these SIP stacks.
Anyone having any doc/link that compares these two SIP servers.
The major criteria that we are looking for are:
- Simplicity of using the SIP stack (Most IMP)
- Perfomance
- Support for Presence
Any pointers in this direction will be highly appreciated.
Thanks,
Pankaj
The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments.
WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email.
www.wipro.com
Hello everyone!
My setup is as follow:
for local PSTN calls
UA (with G.711 and G.729) - my Openser - my Asterisk B2BUA for prepaid -
PSTN gateway (G.711 codec only)
for international calls
UA (with G.711 and G.729) - my Openser - my Asterisk B2BUA for prepaid -
ITSP gateway (G.729 codec only)
B2BUA on the second leg (B2BUA - gateway) always negotiate the right codec, so
no problem, but on the first leg (UA - B2BUA) it always negotiate codec that
is more prefered by the client UA side. So we can get situation, when the
codecs are different on the both legs.
Inspired by the
http://lists.iptel.org/pipermail/serusers/2006-August/030129.html
I decided to normalize codec preference.
Sorry for such a long introduction, at last here goes my problem.
I have the following code in my openser.cfg in INVITE section
# INVITE section
route[3] {
...
if ( route to G.711 capable gateway) {
subst_body("/^m=audio(.*)RTP\/AVP(.*) 8 (.*)/m=audio\1RTP\/AVP 8 \2 \3/ig");
....
}
...
}
and it works like a charm for non-nated clients with real IPs. But when we
should fire use_media_proxy() for a nated one client we get the problem.
SDP body before our substitution (UA-Openser):
v=0
o=20000 12893 12893 IN IP4 192.168.2.134
s=ATA186 Call
c=IN IP4 192.168.2.134
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
After (Openser - B2BUA):
v=0
o=20000 12893 12893 IN IP4 192.168.2.134
s=ATA186 Call
c=IN IP4 217.77.208.184
t=0 0
m=audio 16384 RTP/AVP 8 0 18 101\01560136\012a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
So our subst_body() rearrange codecs as expected, but use_media_proxy() (it
her fault, I think) inserted the mediaproxy fixed port number 60136 between
symbols with ASCII codes \015 CR and \012 LF. This behavior was accepted by
stable Openser 1.1 and latest cvs 1.2.
So the question is :
What to do and who is guilty ? :-)
--
Best regards,
Oleg A. Chernov
OAC4-RIPE
WildPark ISP, Nikolaev, Ukraine
+380512 470555
+380512 500314
Hello List,
I am working on an assignment to develop a load balancing asterisk
cluster. I started with google and found out that SER+ASTERISK is the
best way to do it.
Here is what I am trying to do.
I have a sip account / DID (18xx xxx xxxx ) from an abc company. Now the
company is gateway for between my asterisk server and PSTN network. My
Asterisk server registers with abc company's gateway to receive and make
calls.
How can I do load balancing in this case.
I understand that my SER server should register with the gateway, the
gateway should forward call to SER server and then the server should
forward call to one of the asterisk servers.
Is it possible have the SER server register with another SER or Asterisk
server and act like a client to receive and make calls? Can some one
please point me to some documentation or case study?
Thank you,
-JK
Hi all,
A merry christmas and a great new year to all of you.
I am trying to implement LCR in my openser 1.1. My problem is that I have around 40,000 prefix patterns which i need to enter into the LCR table.
When I populate my LCR table and try to start my openser it dies. I can see the following messages in my log:
convert_row: No memory left
convert_rows: Error while converting row #9055
convert_result: Error while converting rows
store_result: Error while converting result
lcr_reload_gws(): Failed to query lcr data
ERROR: lcr:mod_init(): failed to reload gateways and routes
init_mod(): Error while initializing module lcr
The openser starts properly when I tested with 1000 rows in the LCR table but with 40,000 rows it just dies. Can anyone please help me overcome this issue.
Is there any parameter available to make openser properly run with these many rows in the LCR table.
I am using the LCR in db_only mode. I have not tried this in caching mode. Will the caching mode work with these many rows??
w/regards,
jayesh.
Send free SMS to your Friends on Mobile from your Yahoo! Messenger. Download Now! http://messenger.yahoo.com/download.php
Could anybody pls brief the use of Flex and Bison with SER.
_________________________________________________________________
Christmas is drawing near and the excitement is in the air! Check out this
cool photofeature on Christmas celebrations the world over
http://www.msnspecials.in/photogallery/photogallery.asp?id=a200611814137906
Hi Users;
I am new to use openser so having some problems. I
have installed openser-1.0.0.I have the task to do
load test of openser through SIPp with AUTHENTICATION.
I have added some users in the subscriber table of
openser. Now the problem is how to set the openser.cfg
so that it can response for INVITE and REGISTER
request both like:
The flow goes like:
1)SIPp UAC send REGISTER -------> OPENSER
2)407 Proxy Authentication required <------ OPENSER
3)SIPp send REGISTERwith
credential------>OPENSER(check from subscrible
tables and on the bases of that allow or disallow)
hope someone help me;
Thanks and Regards;
Imran
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
