Could you please upgrade to latest 1.1.0 (cvs update and copy the
mediaproxy directory to stable tree and dload latest mediaproxy server
software from
http://mediaproxy.ag-projects.com/mediaproxy-1.1.0.tar.gz.
Several bugs have been fixed in this release, your problem might be
among those fixed.
Adrian
>>
Hello List,
I have the following setup -
kphoneA------------- [SER A] ---------------- [SER B] ---------- kphoneB
I am forcing both SER A and B to use mediaproxy regardless of whether
the UA's are natted or not (ser.cfg file attached).
When kphoneA makes a call to kphoneB, use_media_proxy() works fine on
SER A. But on SER B, it fails (I can see an error - mediaproxy: Call-Id
not found - but ethereal traffic dump shows that the Call-Id is present
in the SIP message). So I get a call between khponeA---serA---kphoneB.
When I initiate the call from kphoneB, the whole thing reverses and it
becomes kphoneB---serB----kphoneA.
I am using ser-0.8.12 (cannot use cvs version at the moment due to some
code changed we have done) and mediaproxy-1.0.1.
Anyone got any clue where I am wrong? ser.cfg file attached. Any help
will be greately appreciated.
Regards,
Dhiraj
Hello List,
I have the following setup -
kphoneA------------- [SER A] ---------------- [SER B] ---------- kphoneB
I am forcing both SER A and B to use mediaproxy regardless of whether the UA's are natted or not (ser.cfg file attached).
When kphoneA makes a call to kphoneB, use_media_proxy() works fine on SER A. But on SER B, it fails (I can see an error - mediaproxy: Call-Id not found - but ethereal traffic dump shows that the Call-Id is present in the SIP message). So I get a call between khponeA---serA---kphoneB.
When I initiate the call from kphoneB, the whole thing reverses and it becomes kphoneB---serB----kphoneA.
I am using ser-0.8.12 (cannot use cvs version at the moment due to some code changed we have done) and mediaproxy-1.0.1.
Anyone got any clue where I am wrong? ser.cfg file attached. Any help will be greately appreciated.
Regards,
Dhiraj
---------------------- ser.cfg ----------------------------------
-------------------
#
# $Id$
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
debug=9 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=yes # (cmd line: -E)
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=2
fifo="/tmp/ser_fifo"
sip_warning=yes
# ------------------ module loading ----------------------------------
# set the proper path to your modules modules
# if you did "make install" it must be, for example
# loadmodule "/usr/local/lib/ser/modules/sl.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/textops.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"
# Uncomment this if you want to use SQL database
#loadmodule "../sip_router/modules/mysql/mysql.so"
loadmodule "/usr/local/lib/ser/modules/domain.so"
loadmodule "/usr/local/lib/ser/modules/dbtext.so"
loadmodule "/usr/local/lib/ser/modules/mediaproxy.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# modparam("usrloc", "db_mode", 2)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
# modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser")
# -- auth params --
# -- mediaproxy params --
modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock")
modparam("mediaproxy", "sip_asymmetrics", "/etc/ser/sip-asymmetrics-clients")
modparam("mediaproxy", "rtp_asymmetrics", "/etc/ser/rtp-asymmetrics-clients")
modparam("mediaproxy", "natping_interval", 20)
modparam("domain", "db_url", "/usr/local/ser/domaintables")
modparam("domain", "domain_table", "domain")
modparam("domain", "domain_col", "domain")
# ------------------------- request routing logic -------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwars==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;
};
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
if (client_nat_test("3")) {
if (method == "REGISTER" || ! search("^Record-Route:")) {
fix_contact();
force_rport();
};
};
if (uri==myself) {
if (method=="REGISTER") {
log("REGISTER");
save("location");
break;
};
lookup("location");
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;
};
};
if (method=="INVITE") {
use_media_proxy();
t_on_reply("1");
};
if (method == "BYE" || method == "CANCEL") {
end_media_session();
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
break;
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
t_on_reply("1");
if (!t_relay()) {
sl_reply_error();
};
}
onreply_route[1] {
if (status=~"(183)|2[0-9][0-9]") {
if (client_nat_test("1")) {
fix_contact();
};
use_media_proxy();
};
if (status=~"[3-4]0[0-9]") {
end_media_session();
break;
};
}
Hi all,
i have configured SER to manage internal SIP calls between users with same realm (nocable.it). To allow users to do normal calls to fixed phones in Italy, i have to configure the SER in proxy mode to route the calls to the PSTN gateway.
How can i do it?
Thanx a lot
Francesco Russo
Hi all,
i know this a bit OT on this list but nevertheless:
Has anyone experience with the new "1.0.5.3" firmware release
for the Grandstream BT Line fone types? We have a kind of
"rapid registration" problem with a certain firmware release
and so far the actuall firmware doesnt seem to have this bug.
As i am usually using my trusty sipura adapter and not a GS fone
i wanted to ask if someone has made any bad experiences with
the current firmware release before upgrading a batch of those
thingies to the current release.
regards,
:: arnd ::
Hi list!
I just came across something very strange when using the radius-modules
and wonder if it is a wanted feature, a bug or simply me being stupid
(which I guess will be the case).
The thing is the following. My ser.cfg has the following in it when an
UA registers:
if (method=="REGISTER") {
if (!radius_proxy_authorize("XXX.XXX.XXX.XXX"))
{
proxy_challenge("XXX.XXX.XXX.XXX", "0");
break;
};
log(1,"Registered");
save("location");
break;
};
This works fine, means the user get's registered, if it is known to
Radius and not registered in the opposite case.
Now to the strange thing. In most UAs you can enter different user-parts
of the URI and Authentication-Users. I used kphone for this test and
entered a valid username as authentication username and some random
number (or word, that doesn't matter) as "User part of SIP URL". What
happens then is, that the user can register and gets a URI different
from the authenticated username. With this behavior every user would be
able to "hijack" connections from other user.
How can I tell SER to not allow this? Has it something to do with the
SIP-Rpid argument in Radius? Ser seems to ignore it.
Any hints, or RTFMs to get me looking in the right direction to solve
this problem would be very kind.
Best regards
Kai
--
Kai Militzer WESTEND GmbH | Internet-Business-Provider
Technik CISCO Systems Partner - Authorized Reseller
Lütticher Straße 10 Tel 0241/701333-11
km(a)westend.com D-52064 Aachen Fax 0241/911879
Dear Friends,
i have installed SER correctly, it works well and registers users present in mysql database.
After SER installation i have installed SERWEB following step by step Dan Austin's HOW-TO.
I have encountered this problems:
1) The web interface don't displays correclty...i don't see the background images, but the buttons are visible.
2) When a user logs in, on the missed_calls tab, i see the message "error in SQL query, line: 77"
3) When i click on email confirmation link....i see on the web page the message "400 Table "aliases" not in memory, please run save("aliases") or lookup("aliases") in the confirmation script first"
Can you help me to solve this problem, please?
Thanks all in advance
Francesco Russo
Hello,
--- Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote:
> Hi,
<snip>
> Summarizing, only softphones and end-point applications will initiate
> requests, that will be forwarded by servers within the "backbone" to the
> appropriate end-point.
>
> I hope it is more clear now,
Very clear! Thanks a lot for your reply...
> Samuel.
Regards,
=====
Girish Gopinath <gr_sh2003(a)yahoo.com>
__________________________________
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Hi,
SIP's main entity are User Agents (UAs), which can be devided into UAC
(User Agent Client) and UAS (User Agent Server), depending on the
*transaction* they are involved into.
UAC are transaction-level entities initiating requests and accepting
responses. On the other side, UAS are transaction-level entities
accepting requests and issuing responses.
Thus, a normal SIP softphone instantiates (depending on the
implementation may differ the exact way) a UAC whenever sending an
INVITE and the *same* UAC waits for the response (usually an OK).
If acting as receiving point, it will create a UAS instantiation as
soon as it receives an incoming INVITE and will issue the appropriate
answer.
The point is that a server is mainly a UAS waiting for incoming
requests BUT it will need to instantiate a UAC to send requests (just as
forwarding INVITES, ACK, CANCEL,etc...) only after receiving incoming
requests. By itself, a server MUST NOT send a request except to forward
incoming ones.
Next diagram (I expect it arrives with the right alignment) tries to
explain the transaction-level entities involved in a generic SIP
transaction. (INVITE+OK). I hope it will be useful to clarify how SIP
works.
softphone proxy
server softphone
UAC----------------------------------------->UAS +
UAC----------------------------------------->UAS
Thus a SIP application starting a request (initiating a transaction)
will create a UAC transaction and will expect a UAS to receive and
process it.
I expect not to confuse you with the words UAS and UAC with the usual
meaning of UA (=softphone).
Summarizing, only softphones and end-point applications will initiate
requests, that will be forwarded by servers within the "backbone" to the
appropriate end-point.
I hope it is more clear now,
Samuel.
P.S. This is the normal SIP behaviour........I'm afraid some situations
might be different (NOTIFY requests).
Unclassified
>>> GR S <gr_sh2003(a)yahoo.com> 07/08/04 09:00AM >>>
Hi,
--- Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote:
> So maybe you should start with your intention - what feature are you
> trying to setup?
> ser can create requests ( transaction = request+response(s) ) and you
> can use this feature via the FIFO.
Thanks for the reply.
Sorry, I misnderstood the meaning of transaction. I was only thinking
about SIP requests, not
responses. But i am still confused about call initiating. Please
correct me if i am wrong on this,
the original poster was asking about initiating calls from SER and
AFAIK proxies dont initiate
calls. Can they initiate calls? If yes, on what situations? When you
say SER can create requests,
do you mean methods like CANCEL?
> klaus
TIA,
=====
Girish Gopinath <gr_sh2003(a)yahoo.com>
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