Hi,
Can I know the Call ID of a UAC / UAS who has been registered on th eproxy server?
I need to send the MESSAGE to the Particular "UAC" but I do not know the Call ID ?
There is no INVITE nor Call Established.But I need to send IM to that particular UAC.
As Far As I Know, Call id is mandatory , Hence I need to know the Call id from the prosy server.
How should I proceed?
Thanks and Regards.
Abhijit
hi bogdan,
Thanx for the help. I am sorry to ask again but how do i use the "trusted_reload" command in the script. I tried different techniques but i could not get the results.
please help!!
Thanx again
Hi all.
I have configured ProxySER for my test clients and connected they via
SER, but... I can't configure SER - MySQL interoperability. I've read
"SIP.edu Cookbook" by ProxySER afterward I made the changes into ser.cfg
and db configuration. After SER startup I've got the next log:
Aug 2 19:21:21 fwd ser[11846]: Maxfwd module- initializing
Aug 2 19:21:21 fwd ser[11849]: new_connection(): Access denied for user
'serro'@'localhost' (using password: YES)
Aug 2 19:21:21 fwd ser[11849]: db_init(): Could not create a connection
Aug 2 19:21:21 fwd ser[11849]: ERROR: auth_db_bind: unable to connect
to the database
Aug 2 19:21:21 fwd ser[11849]: init_mod_child(): Error while
initializing module auth_db
Aug 2 19:21:21 fwd ser[11849]: ERROR: open_uac_fifo: init_child failed
Aug 2 19:21:21 fwd ser[11849]: starting fifo server failed
I've had the next config:
#
# $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
# debug=7 # debug level (cmd line: -dddddddddd)
# fork=yes
# log_stderror=yes # (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"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "//lib/ser/modules/mysql.so"
loadmodule "//lib/ser/modules/sl.so"
loadmodule "//lib/ser/modules/tm.so"
loadmodule "//lib/ser/modules/rr.so"
loadmodule "//lib/ser/modules/maxfwd.so"
loadmodule "//lib/ser/modules/usrloc.so"
loadmodule "//lib/ser/modules/registrar.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "//lib/ser/modules/auth.so"
loadmodule "//lib/ser/modules/auth_db.so"
loadmodule "//lib/ser/modules/acc.so"
loadmodule "//lib/ser/modules/exec.so"
# loadmodule "//lib/ser/modules/group.so"
# loadmodule "//lib/ser/modules/msilo.so"
loadmodule "//lib/ser/modules/print.so"
loadmodule "//lib/ser/modules/textops.so"
loadmodule "//lib/ser/modules/uri.so"
# loadmodule "//lib/ser/modules/uri_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)
modparam("usrloc", "db_url", "mysql://ser:mypass@fwd.serout.com/myserdb")
modparam("auth", "secret", "myphrase")
# -- 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 -------------------
skipped...
Where I can dig for solving this trouble?
Thank you in advance.
Alexander Naumov.
Hi All,
Does anyone knows Billing Server for SIP based on SER ? whether for
prepaid or postpaid system.
I would be glad if there's Opensource SER Billing system but Licensed
Billing also ok for me.
Thanks & Regards,
Asep
Just one remark: only one of the two URI will be translated to the contact.
I'll try to explain...
imagine the incoming Req-URI is 1001(a)test.com. When it passes the first lookup,
lookup("aliases");
SER will retrieve the two aliases and put one of them as the new req-uri and the other as branch.
Req-URI: 1002(a)test.com
branch: 1003(a)test.com
Doing the second lookup,
lookup("location");
will put the contact of 1002(a)test.com as Req-URI (if more than one contact is present, the subsequent bindings will be added as branches) and the 1003(a)test.com will be left as branch untouched.
Req-URI: 1002(a)10.1.1.102
branch: 1003(a)test.com
This happens because lookup() only takes as input the req-URI.
The result is that SER makes a DNS query for test.com, to send the request to 1003(a)test.com, instead of the end-point. If the DNS is properly configured, test.com should point to SER itself, making SER sending the message to itself via the network. Depending on the config file, it can work but you should check the logs to see what is exactly happening.
User 1002 will get the message without any problem.
NOTE: depending on the order in the database, 1003 and 1002 can exchange roles in upper explanation.
As SER is now coded, above can not be changed (as far as I know)....
Hope it helps,
Samuel.
Unclassified.
>>> "Zeus Ng" <zeus.ng(a)isquare.com.au> 08/03/05 01:51AM >>>
You would need two steps here, looup("aliases"), then lookup("location").
Zeus
> -----Original Message-----
> From: serusers-bounces(a)lists.iptel.org
> [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard Z
> Sent: Wednesday, 3 August 2005 3:33 AM
> To: Atle Samuelsen
> Cc: serusers(a)lists.iptel.org
> Subject: Re: [Serusers] lookup 2 uri
>
>
> Hi Atle,
>
> Alias doesn't replace uri with its real location. I'd like to use
> lookup("location") to find out its real location. For
> example, if 102(a)test.com is at 102(a)10.1.1.102 and
> 103(a)test.com is at 103(a)10.1.1.103. I'd like to replace all
> uris with 102(a)10.1.1.102 and 103(a)10.1.1.103 in a single ser
> transaction logic. Would it be possible?
>
> Thanks,
> Richard
>
>
> On 8/1/05, Atle Samuelsen <clona(a)camaro.no> wrote:
> >
> > Hi,
> >
> > you could do something like :
> >
> > serctl alias add 101 sip:102@test.com
> > serctl alias add 101 sip:103@test.com
> >
> > This way the invite would be forked to both 102 and 103,
> >
> > just remember to have both lookup("aliases"); and
> save("aliases"); in
> > your config at the right spot :-)
> >
> > -Atle
> >
> >
> >
> > * Richard Z <rzheng(a)gmail.com> [050802 04:34]:
> > > Hi,
> > >
> > > I am trying to implement a feature to ring two ip phones
> > > simultaneously. For example, when I call 101(a)test.com, it
> rings both
> > > 101(a)test.com and 102(a)test.com. In this case, the ruri is
> > > 101(a)test.com. I can use append_branch 102(a)test.com, but
> is there a
> > > way to lookup the locations of both 101 and 102?
> > >
> > > Thanks,
> > > Richard
> > >
> > > _______________________________________________
> > > Serusers mailing list
> > > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
> > >
> >
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
>
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Hi, there
In INSTALL doc [http://www.openser.org/docs/INSTALL] section 3-C, it seems
to me that a gentoo's ebuild is already available in the official portage
tree for OpenSER. Unfortunately I couldn't find it in my daily-rsync'ed
Gentoo box, nor I can find it in http://packages.gentoo.org.
Searching in both users and devel mailing lists and Google search couldn't
find anything useful either.
Am I missing something obvious here? Could someone give me a hint please?
Thanks.
-- Cheng
I'm using nathelper with SER. Here's a quickie diagram of my
setup:
[SIP ATA] <-------------> [SER] <------------------> [Asterisk]
10.0.201.5 eth1 - 10.0.201.7 eth0 - <public_ip1>
eth0 - <public_ip0>
I'm using the basic nathelper example configuration, except for one
modification. I removed the call to t_relay and replaced it with:
t_relay_to("<public_ip1>", "5060");
Basically I want a SIP packet to come in to eth1 on SER server, get
rewritten as needed by nathelper, then get sent out to public_ip1 on my
Asterisk PBX.
This appears to be working except a tcpdump on eth0 on SER shows:
14:39:08.654908 IP 10.0.201.7.5060 > <public_ip1>.5060: UDP, length 629
Obviously the packets will never get back with that source IP address. What
command should I be looking at to rewrite this IP so it shows up as
<public_ip0> automatically?
I'm trying to get SIP ATA's to REGISTER on my Asterisk box but proxy through
SER ... (and eventually make use of the rtp proxy).
Can post more info if requested, I am new to SER. :-)
--
Ray Van Dolson
Linux/Unix Systems Administrator
Digital Path, Inc.
Hi all:
I am using ser 0.8.14 with rtpproxy and kphone 0.4.1.1,
When kphone holds the call, then unholds later,
sometimes kphone will change the rtp port in SDP differ from previously
used.
Therefore, the rtp packet can't be forward correctly via rtpproxy if ser
doesn't signal rtpproxy to change the forwarding port mapping
I have asked before about this problem, and Zeus Ng have give me some
information
that the key factor is to catch the re-invite, but i don't know how to
verify if a INVITE is really a INVITE or r reINVITE...
Is there any method can let ser catch or verify the re-invite message in
ser.cfg ?
Thanks and regards
Jimmy
Please help.....
----- Original Message -----
From: Nicky
To: sems(a)lists.iptel.org ; serusers(a)lists.iptel.org
Sent: Tuesday, July 05, 2005 6:56 PM
Subject: Problem on pthread and dtmf
Dear all,
Anyone encountered the problem below,please help. Thank you.
----- Original Message -----
From: Nicky
To: sems(a)lists.iptel.org ; serusers(a)lists.iptel.org
Sent: Thursday, June 16, 2005 3:51 PM
Subject: Problem on pthread and dtmf
Dear all,
1. Please help, when i run sems -E i experienced this error, what can be the caused of it:~
~~11352) ERROR: start (AmThread.cpp:73): pthread create failed with code 12
2. There is signal of DTMF which is not pressed but is being send to the server, please see below for the log. Please advice what can be the caused and problem.
Thank you in advance.
~~(4395) DEBUG: isdn_audio_eval_dtmf_relative (IvrDtmfDetector.cpp:349): Posting D
TMF 0: 5 (5)
(4395) DEBUG: postEvent (AmEventQueue.cpp:47): AmEventQueue: trying to post even
t
(4395) DEBUG: postEvent (AmEventQueue.cpp:55): AmEventQueue: event posted
(4395) DEBUG: processEvents (AmEventQueue.cpp:68): before processing event
(4395) DEBUG: process (IvrPython.cpp:988): IvrPython processing event...
(4395) DEBUG: process (IvrPython.cpp:991): IvrDialog processing event...
(4395) DEBUG: onDTMFEvent (IvrPython.cpp:1130): IvrPython::onDTMFEvent(): callin
g onDTMFCallback key is 5...
(4395) DEBUG: ivrEmptyMediaQueue (IvrPython.cpp:102): IVR: emptying media queue
.
(4395) DEBUG: IvrMediaEvent (IvrEvents.cpp:31): New Media Event: 1,
(4395) DEBUG: postEvent (AmEventQueue.cpp:47): AmEventQueue: trying to post even
t
(4395) DEBUG: postEvent (AmEventQueue.cpp:55): AmEventQueue: event posted
(4395) DEBUG: ivrStopRecording (IvrPython.cpp:132): IVR: stop recording.
best regards,
nicky
Method==Bye ?
Try a ngrep during a call transaction to see what are the header sent.
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Anderson Alves de Albuquerque
Sent: Wednesday, August 03, 2005 12:20 AM
To: serusers(a)lists.iptel.org
Subject: [Serusers] Start and fnish call
With SER how can I know when the client SIP finish a call?
I need that SER can start a script when the client SIP finish a call.
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