Hello,
A little green here, looking for guidance.
I have a simple UA server that can respond to invites. It will respond with
a 302 (temp. moved) to free up 5060 for the next invite. It uses only TCP
for RTP transport.
I am trying to use OpenSER with RTPProxy to be a, well, RTP proxy.
The SIP trunk providers will map the DIDs to the OpenSER server, passing UDP
for RTP.
I'm assuming this is rather easy, but I'm having trouble getting through the
documentation.
I currently have OpenSER 1.3 and RTPProxy running.
A step-by-step list of openserctl or module commands (vague is fine, I can
read up on it) would be greatly appreciated. So would any other advice -
anything more than RTFM ;-)
Thanks!
Eric
Hello,
A little green here, looking for guidance.
I have a simple UA server that can respond to invites. It will respond with
a 302 (temp. moved) to free up 5060 for the next invite. It uses only TCP
for RTP transport.
I am trying to use OpenSER with RTPProxy to be a, well, RTP proxy.
The SIP trunk providers will map the DIDs to the OpenSER server, passing UDP
for RTP.
I'm assuming this is rather easy, but I'm having trouble getting through the
documentation.
A step-by-step list of openserctl commands would be greatly appreciated. So
would any other advice.
Thanks!
Eric
I have a trouble in my serweb instalation, I defined a virtual host "
sip.mydomain.com" but when I am accessing via http, its logon like @
mydomain.com , and the users via openserctl add have the host on the domain
in the table suscribers. how can I correct this ?
--
Thanks a lot
Frank Gonzalez
414-6260492
--
Saludos.
Frank Gonzalez
414-6260492
Hi All
When I logon using usernames with letters all is fine when I logon using
usernames all numbers I get the following error and the UA does not
logon
Registration Error: 501 Not Implemented
This is my config
http://pastebin.com/md7d0de4
It used to work with this config
http://pastebin.com/m10e981d2
Thx for any suggestions.
Hello,
Im coding a module for my needs, and one of them is to
modify the SIP message after querying some db.
I decided to use the function replace_all() exported
by the textops module, using the find_export()
function declared in sr_module.h.
The problem is that im getting a SIGSEGV when some
child exec the replace_all mapped function:
Feb 11 11:29:30 [4868] DBG:core:find_cmd_export_t:
found <replace_all>(2) in module textops
[/usr/local/lib/openser/modules/]
Feb 11 11:29:31 [4867] INFO:core:handle_sigs: child
process 4868 exited by a signal 11
Feb 11 11:29:31 [4867] INFO:core:handle_sigs: core was
generated
Feb 11 11:29:31 [4867] INFO:core:handle_sigs:
terminating due to SIGCHLD
I cant find the core generated by openser. Where is
the default dir where this cores are saved?
I dont know if this is a correct way to modify a SIP
message; is there a better o more apropiated way to do
this?
Thanks in advance, Francisco.
Tarjeta de crédito Yahoo! de Banco Supervielle.
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Hi Iñaki,
thanks alot for the pointer, did a quick search for onreply_route and
found this article:
http://www.voip-info.org/wiki/view/OpenSER+And+Mediaproxy
By using the route from this example and fixing it to use mediaproxy in all
situations I finally have my RTP routing via the mediaproxy box! :D I still
have to read up on the onreply_option to fully understand it.
Perhaps if this is something that is always required referenec to it should
be added
to the mediaproxy module documentation??
Thanks again! Cheers Andy.
> I think you have not configured the "onreply_route" so SDP is not modified
in
180/183/200 reply, so MediaProxy doesn't work at all.
________________________________________________
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Hi All
I was wondering about the rtpproxy and mediaproxy modules, if I do use
either of them to allow clients behind NAT to do VOIP calls does that
mean that I do not need to use a STUN server anymore ?
Or should I use both a STUN server and the rtp proxy ?
What would provide the best solution in terms of solving NAT related
problems ?
Thanks
Hi, I set:
-------------------------------------------
dst_blacklist = gw:{( any , GW_IP, 0 , "" )}
[...]
t_on_branch("ON_BRANCH_TO_USER");
if (!lookup("location")) {
if ($rc == -5)
xlog("L_WARN", "WARN: Fraudulent Contact !!! \n");
sl_send_reply("404", "Not Registered");
exit;
}
t_relay();
[...]
branch_route[ON_BRANCH_TO_USER] {
use_blacklist("gw");
}
-------------------------------------------
I do a fraudulent REGISTER for an AoR (with GW_IP in "Contact" header) and
when calling that AoR I get a correct "473 Filtered destination". But I don't
see the WARN, so "lookup(location)" didn't reply -5.
Why not? the IP in Contact column for taht AoR is GW_IP,
why "lookup(location)" doesn't realize of it?
It supposed that "lookup()" returns -5 if the destination is filtered (black
listed):
http://www.openser.org/docs/modules/1.3.x/tm.html#AEN336
Maybe it's a bug?
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es