Is there a module being develop for ser that give it b2bua capabilities
or is there a b2bua that integrates well with openser?
Pablo Delgado
Digiport Miami LLC
Phone: (305) 424-0016
Web: www.digiportmiami.com
Email: pablo(a)digiportmiami.com
-----Original Message-----
From: Juha Heinanen [mailto:jh@tutpro.com]
Sent: Tuesday, June 20, 2006 4:01 PM
To: Pablo Delgado
Subject: RE: [Users] Privacy (*67)
Pablo Delgado writes:
> To other sip phones.
then you need a b2bua that terminates the call and creates another one,
where caller has been changed to anonymous.
-- juha
Bogdan,
I upgraded to the current cvs version, and I think that has fixed it. With no change to openser.cfg, it's been running for about 3 hours now without any reported memory issues.
Douglas.
> -----Original Message-----
> From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro]
> Sent: Monday, June 26, 2006 10:20 AM
> To: Douglas Garstang
> Cc: Users(a)openser.org
> Subject: Re: [Users] Out of Memory
>
>
> Ok Douglas - please let me know if the latest changes in
> resolver have
> any effect on the leak you observed.
>
> regards,
> bogdan
>
> Douglas Garstang wrote:
>
> >Bogdan,
> >
> >Don't appear to have ever received that message.
> >
> >To answer your questions:
> >We are not using ENUM at all. We aren't using our own module
> for DNS lookups.
> >I will try the latest cvs.
> >
> >Doug
> >
> >
> >
> >>-----Original Message-----
> >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro]
> >>Sent: Monday, June 26, 2006 8:33 AM
> >>To: Douglas Garstang
> >>Cc: Users(a)openser.org
> >>Subject: Re: [Users] Out of Memory
> >>
> >>
> >>See the list. I sent you a reply :
> >> http://www.openser.org/pipermail/users/2006-June/005321.html
> >>
> >>regards,
> >>bogdan
> >>
> >>Douglas Garstang wrote:
> >>
> >>
> >>
> >>>>-----Original Message-----
> >>>>From: Douglas Garstang
> >>>>Sent: Thursday, June 22, 2006 12:09 PM
> >>>>To: Bogdan-Andrei Iancu
> >>>>Cc: Users(a)openser.org
> >>>>Subject: RE: [Users] Out of Memory
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>>-----Original Message-----
> >>>>>From: Douglas Garstang
> >>>>>Sent: Wednesday, June 21, 2006 12:48 PM
> >>>>>To: Bogdan-Andrei Iancu
> >>>>>Cc: Users(a)openser.org
> >>>>>Subject: RE: [Users] Out of Memory
> >>>>>
> >>>>>
> >>>>>Bogdan,
> >>>>>
> >>>>>I finally managed to upload the memory dumps to pastebin.
> >>>>>
> >>>>>
> >>Links are:
> >>
> >>
> >>>>>Hopefully you can access them.
> >>>>>
> >>>>>Memory dump right after OpenSER was started:
> >>>>>http://pastebin.com/723872
> >>>>>Memory dump after OpenSER running for 20min:
> >>>>>http://pastebin.com/723890
> >>>>>Memory dump after OpenSER running for 50min:
> >>>>>http://pastebin.com/723902
> >>>>>
> >>>>>The problem started to occur between the 20 and 50 minute
> >>>>>samples. About 20 calls had been processed in that time.
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>So... I'm wondering if anything has been determined with this...
> >>>>
> >>>>
> >>>>
> >>>>
> >>>Hi Bogdan. Did you manage to find anything with this?
> >>>
> >>>Douglas.
> >>>
> >>>
> >>>
> >>>
> >
> >
> >
>
>
Hello,
I am using SER-0.9.6. How to send instant message programmatically, how
to build sip message and which function can by used(like t_uac,
t_request).
Can you help where and which to be changed in ser code.
Thank you,
Regards,
Sriram Srinivas.
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www.wipro.com
Hello,
how could it be done through AVP´s and database lookup, to do a parallel
ringing.
Following setup:
If uri eg example(a)test.dom is called, this call should be forwarded in
parallel to a pstn-gateway.
That means, receive the AVP with the telephonnumber associated to uri
example(a)test.dom, rewrite uri, rewrite host/port and forward in parallel
to gateway.
I tried "append_branch", but was not able to put in the AVP
(phonenumber), rewrite host/port points directly to the gateway.
I have no troubles to load the AVP, only to make is possible, that the
SIP Client and PSTN Phone are ringing in parallel.
Some suggestions ?
regards,
Andreas
Okay, I'm trying to use uac_replace_from to anonymize the From: in order to
suppress ANI. I think I've got it, but it seems like the behavior isn't what
I expect. First, the relevent config file section:
if (search("Anonymous")) {
uac_replace_from("","sip:anonymous@xxxxxxx.com");
append_rpid_hf("",";party=calling;id-type=subscriber;screen=yes;privacy=full");
log(1,"Made a call anon");
};
Now, the headers as they go in and come out:
SIP MESSAGE 4 xx.7.96.185:5061() -> xx.7.96.82:5060()
UDP Frame 4 30/Jun/06 08:30:57.4723
TimeFromPreviousSipFrame=0.0069 TimeFromStart=0.0204
INVITE sip:1360789xxxx@voip02.xxxxxxx.com SIP/2.0
Via: SIP/2.0/UDP xx.7.96.185:5061;branch=z9hG4bK-7fa6fcd6;rport
From: Anonymous
<sip:+136091xxxxxx@voip02.xxxxxxxx.com>;tag=8c07328919ade13ao1
To: <sip:1360789xxxx@voip02.xxxxxxxx.com>
Call-ID: 51835bf5-ba1be6be@localhost
CSeq: 102 INVITE
Max-Forwards: 70
Contact: Anonymous <sip:+136091xxxxx@xx.7.96.185:5061>
Expires: 240
User-Agent: Sipura/SPA2000-3.1.5
Content-Length: 311
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
SIP MESSAGE 6 xx.7.96.82:5060() -> xx.7.96.90:5060()
UDP Frame 6 30/Jun/06 08:30:57.4791
TimeFromPreviousSipFrame=0.0007 TimeFromStart=0.0272
INVITE sip:360789xxxx@xx.7.96.90:5060 SIP/2.0
Record-Route: <sip:xx.7.96.82;ftag=8c07328919ade13ao1;lr=on>
Via: SIP/2.0/UDP xx.7.96.82;branch=z9hG4bKe9d1.8c515a97.0
Via: SIP/2.0/UDP xx.7.96.185:5061;branch=z9hG4bK-7fa6fcd6;rport=5061
From: Anonymous <sip:36091xxxxx@voip02.xxxxxxxx.com>;tag=8c07328919ade13ao1
sip:anonymous@xxxxxxxx.com
To: <sip:360789xxxx@xx.7.96.90:5060>
Call-ID: 51835bf5-ba1be6be@localhost
CSeq: 102 INVITE
Max-Forwards: 69
Contact: Anonymous <sip:+136091xxxxx@xx.7.96.185:5061>
Expires: 240
User-Agent: Sipura/SPA2000-3.1.5
Content-Length: 311
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
Remote-Party-ID: "xxxxxxxx Customer"
<sip:+136091xxxxx@xxxxxxxx.com>;party=calling;id-type=subscriber;screen=yes;privacy=full
And finally what I'm seeing in the log:
Jun 30 08:18:33 voip02 /usr/sbin/openser[4684]: DEBUG:uac:replace_from:
removing display [Anonymous]
Jun 30 08:18:33 voip02 /usr/sbin/openser[4684]: DEBUG:uac:replace_from: uri
to replace [sip:+136091xxxxx@voip02.xxxxxxxx.com]
Jun 30 08:18:33 voip02 /usr/sbin/openser[4684]: DEBUG:uac:replace_from:
replacement uri is [sip:anonymous@xxxxxxxx.com]
So what am I missing here? It looks like uac_replace_from is finding the
right URI and replacing it, just not properly. I've already gone into CVS
and I don't see any changes since 1.0.1 (which is what is running here), nor
any relevent changes from SER compared to OpenSER.
Any help? Please?
-Keith
Hi Joao,
No I was not able to solve the issue.
It seems (this is my guess tough) that the Portaone RTP proxy assumes that it has one public IP adress, so the valid configuration to use it is Public Nt-Private Nt. I was not able to make it work in other configurations (neiher I got feedback from Portaone to do so).
Nevertheless the code is available, so it could be modified...as long as you have the time and will to do so. I did not ;).
Best regards,
josé
-----Original Message-----
From: Joao Pereira [mailto:joao.pereira@fccn.pt]
Sent: 19. oktober 2005 20:17
To: Jose Soler; serusers(a)lists.iptel.org
Subject: Re: [Serusers] RTP proxy between two subnetworks with private @s
Hello, did you made it to put the clients of networks A and B to call
each other?
I want to do the same, and tried a lot of SER/RTPproxy configurations,
including the one in: /ser-0.9.0/modules/nathelper/examples/alg.cfg
and also tried to run rtpproxy with the "-l 10.0.0.135/193.136.2.2" option. But I just was able to ring the phones (wen calling between networks),
but the RTP doesnt pass...
If you found the solution, please tell me.
Thanks
Joao Pereia
www.fccn.pt
Jose Soler wrote:
> Hi,
>
> I am trying to figure out how to solve the follwoing problem. I have
> two subnetworks, A and B, with different private ip adressing schemes
> (IP@A <mailto:IP@A>) and (IP@B <mailto:IP@B>).
>
> SER is installed in a computer with network interfaces towards both
> subnetworks.
> SER's SIP signalling proxying operation works properly within the
> subnetworks and when trying to set up a communication between users in
> A and B. But in that last case, obviously there is no media at all
> circulating among the subnetworks.
>
> Portaone's RTP proxy has been installed and configured in the computer
> with interfaces towards both subnetworks where SER is installed.
>
> I am trying to configure SER so that, based on the nathelper module,
> when communication between both subnetworks occurs, the RTP proxy is
> involved and the communication (also media and not only signalling) is
> possible. BUT I am making something wrong, becouse it does not work ...
>
> Can anyone give me a hand /hint?
> Thanks a lot in advance / in any case.
>
> My SER config file is the following:
>
>
> #
>
> # ----------- global configuration parameters ------------------------
>
> /* 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)
>
> fifo="/tmp/ser_fifo"
>
> fifo_mode=0662
>
> alias=wirelessip.x.x.x
>
> alias=sip..x.x.x
>
> alias=x.x.x
>
> log_stderror=no
>
> debug=3
>
> children=3
>
> mhomed=1
>
> # ------------------ 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/textops.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"
>
> # For NAT support / media proxying
>
> loadmodule "/lib/ser/modules/nathelper.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)
>
> # For NAT
>
> # We will use flag 6 to mark NATed contacts
>
> modparam("registrar", "nat_flag", 6)
>
> # Enable NAT pinging
>
> modparam("nathelper", "natping_interval", 60)
>
> # Ping only contacts that are known to be
>
> # behind NAT
>
> modparam("nathelper", "ping_nated_only", 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;
>
> };
>
> # special handling for NATed clients; first, nat test is
>
> # executed: it looks for via!=received and RFC1918 addresses
>
> # in Contact (may fail if line-folding used); also,
>
> # the received test should, if complete, should check all
>
> # vias for presence of received
>
> if (nat_uac_test("3")) {
>
> # allow RR-ed requests, as these may indicate that
>
> # a NAT-enabled proxy takes care of it; unless it is
>
> # a REGISTER
>
> if (method == "REGISTER" || ! search("^Record-Route:")) {
>
> log("LOG: Someone trying to register from private IP, rewriting\n");
>
> # This will work only for user agents that support symmetric
>
> # communication. We tested quite many of them and majority is
>
> # smart smart enough to be symmetric. In some phones, like
>
> # it takes a configuration option. With Cisco 7960, it is
>
> # called NAT_Enable=Yes, with kphone it is called
>
> # "symmetric media" and "symmetric signaling". (The latter
>
> # not part of public released yet.)
>
> fix_nated_contact(); # Rewrite contact with source IP of signalling
>
> if (method == "INVITE") {
>
> fix_nated_sdp("1"); # Add direction=active to SDP
>
> };
>
> force_rport(); # Add rport parameter to topmost Via
>
> setflag(6); # Mark as NATed
>
> };
>
> };
>
> # 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;
>
> };
>
> lookup("aliases");
>
> # 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("com.dtu.dk", "subscriber")) {
>
> www_challenge("com.dtu.dk", "0");
>
> break;
>
> };
>
> save("location");
>
> 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();
>
> };
>
> }
>
> #
>
> # Forcing media relay if necessary
>
> #
>
> route[1] {
>
> #if (uri=~"[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")){
>
> # sl_send_reply("479", "We don't forward to private IP addresses");
>
> # break;
>
> #};
>
> #if (isflagset(6)) {
>
> force_rtp_proxy(); # I force everything through the proxy
>
> t_on_reply("1");
>
> append_hf("P-Behind-NAT: Yes\r\n");
>
> #};
>
> if (!t_relay()) {
>
> sl_reply_error();
>
> break;
>
> };
>
> }
>
> onreply_route[1] {
>
> if (status =~ "(183)|2[0-9][0-9]") {
>
> fix_nated_contact();
>
> force_rtp_proxy();
>
> };
>
> }
>
>
>
>
>
>
>
>
>-----------------------------------------------------------------------
>-
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
>
>
Dear all,
Does anyone know if SER (0.8.14) supports PRACK messages ? I want to know
if it forwards them.I want to add in my configuration if(method==PRACK)
{..} and I want to know if I can do that and if SER allows this
configuration line.
Best regards,
Razvan
Hello!
Just curious: I do remove_hf("Contact") right after fix_nated_contact() (for
whatever reason).
Then the remove_hf() removes anything but the new SIP URI from the message,
leaving e.g.
sip:user@received_ip:port
in front of the headerfield following the Contact. e.g.
Before:
Some-Header: bla
Contact: Name <sip:user@privateip:port>
Authorization: bla
Then:
Some-Header: bla
sip:user@public_ip:portAuthorization: bla
How to remove that Contact: after having it nat_helped and then saved?
br
Walter
Hi friends,
I am new in OpenSER, I want to use it only for SIP Proxying with freeradius.
I made plan to install openser with freeradius on Virtual Server to get only
100 cuncurent calls.
1- Is it possible to install on Virtual Server?
2- Which Codecs or used, because i want to calculate the bandhwidth
according to the codec?
3- What RAM should be used to handel 100 cuncurent calls?
4- It can accept h323-credit-time from radius to control max credit call
time?
I will appriciate for your kind of suggestion.
Regards,
www.Go4Calls.Com
VoIP Forums
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