Hi ALL;
Hi Paul;
About Paul's email on SEP.19 with the subject " ingecting SIP NOTIFY INTO SER FIFO" , I come up with the following question:
Why we use "sipsak" for sending "NOTIFY" message from Asterisk to Ser?? Is it because of Asterisk is not a "b2bua"
Warnest Regards
mohammad
How can I compile SER on Mac OS X? With the package I got instructions
came in order to start SER at startup, but I found nothing about
building. I tried typing "make" but I got a very long list of all
kinds o errors.
Can anyone help?
Thanks
Greetings:
I've built CVS HEAD on OpenBSD/sparc (sun4c) with changes to
Makefile.defs, locking.h and other misc. changes; I will send
the patches to whomever wants them. I have it running on a
Sun IPX (our border router); our UAs reside on private address
spaces which works well if the remote UA is on a public IP --
I have yet to deal with remote UAs behind NAT but have built
and run rtpproxy HEAD with 'ser' and done some initial testing.
I would appreciate testing with low-bandwidth codecs; please call
sip:10@cybertheque.org anytime (I may not be around though).
Regards,
Michael Grigoni
Cybertheque Museum
Laurent,
In your post "rtpproxy+nathelper (0.8.14) +video" on Nov 8th you
mentioned you had made same changes to nathelper in order to support
video. We are using the same scenario (but with the original nathelper)
and we already have audio working, but we want also video. So please,
could you send me this modified version?.
Thanks in advance,
Luciano Bajo
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Hello,
i try to test a leadtech ATA with SER.
when i try to register this agent to the SER with nathelper the ATA always
send the same request, i think that he drop the response from the server.
What is false in the response ?
thanks
Laurent
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<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D460502020-19112004>Hello,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D460502020-19112004></SPAN></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D460502020-19112004>i try to=20
test a leadtech ATA with SER.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D460502020-19112004> when i try to=20
register this agent to the SER with nathelper the =
ATA always=20
send the same request, i think that he drop the response from the=20
server.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D460502020-19112004></SPAN></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D460502020-19112004>What =
is false in the=20
response ?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D460502020-19112004></SPAN></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D460502020-19112004>thanks</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D460502020-19112004></SPAN></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D460502020-19112004>Laurent</SPAN></FONT></DIV></BODY></HTML>
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Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
I would like to play back verbal error messages when users try to make
a call outside of their calling permissions, etc. Would SEMS be the
way to go for this? Does anyone have an example of how this would be
set up?
Dear Sirs:
I'm configuring a pre-paid scenario with a third party
gateway. The GW guys gave me a GW number (a.k.a
Username) and a password, so i'm able to place calls
in the PSTN only if i am registered in my SIP Proxy
with the username the GW guys gave me.
The thing is that i want all my users to be able to
use this channel to reach the PSTN. If I call this way
(sip:01xxxxx@theirdomain.com) i succesfully reach the
PSTN but when i call this other way
(sip:01xxxxx@mydomain.com) their server returns a
(403) Forbidden. I'm aware that this behavior is
correct (due to domain verifications) but i need to
know how could I, in my SIP Proxy, rewrite the "To: "
field in order to write the correct domain when i
place a call like this: (sip:01xxxxx@mydomain.com). I
believe that if I reach a solution in this particular
issue i would be able to let all my subscribers to use
individualy a GW account.
If anyone can give me a clue of how could i reach a
solution or any commets regarding this partucular
issue, i would be really grateful.
Best regards.
Andr�s Parra L.
andresparra(a)ipsofactum.com
Ipsofactum Ltd.
www.ipsofactum.com
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I have a scenario where an asterisk machine is working with 2 network cards
(1 public and 1 private). My IP phones are all on private Ip’s It works
great internally but I cannot forward sip traffic to a voip provider due to
NAT. I tried to install SER on the same machine using different ports so the
asterisk forwards all numbers starting with 00 to SER. SER will be
forwarding these calls to a voip SIP provider. Can anyone provide me with a
sample config for SER cause when ever I try to send a number starting with
00, I get a sip error that the number does not exist.
Regards
Chris
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Anyone have any ideas on this? Would it be best to build an external
script to do this? The problem I'm facing is that this should be as
dynamic or automated as possible. It sounds like at the very least I
will have to reference the username manually somehow.. If that makes
sense.
Matt
-----Original Message-----
From: Matt Schulte
Sent: Wednesday, November 24, 2004 7:33 AM
To: serusers(a)lists.iptel.org
Subject: [Serusers] Multiple URI's behind one username/registration
I'm think that I'm wanting to go above and beyond aliases at this point,
could anyone suggest a way to have multiple URI's attached to one
username? ie: For the sake of having multiple phone lines. User aliases
can of course receive the URI but you're translating it back to the
original registered userid/uri .. Ideas?
I think asterisk spoiled me in this sense because on that
platform you can send any URI to users via digest.. :/ I know this
should be possible, just not sure how difficult it will be. Thanks
Matt
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It's my turn for nat troubles now ;)
I've looked at the examples for nat helper and am trying to setup a rtp
proxy (which I eventually want to be able to scale/get off the ser host).
However, I'm using the attached setup (cut the used parts) but when my
client tries to Register, the ser box sends back the reply to the internal
IP of my UA?
So ser receives a REGISTER message via 1.2.3.4 and replies to 192.168.0.x ?
Kind regards,
E. Versaevel
Dear List,
When I try to call any of my routes from the main routing logic routine, I
get the following warning in my log and the call is not processed as it
should be in the route.
0(26610) WARNING: run_actions: null action list (rec_level=3)
In other words, I have:
route{
:
:
:
log(1, "LOG: Calling route 2\n");
route(2);
log(1, "LOG: Done calling route 2\n");
break;
}
route[2] {
log(1, "LOG: I am in Route 2\n");
:
:
:
}
My output is:
0(26841) LOG: Calling Route 2
0(26841) WARNING: run_actions: null action list (rec_level=3)
0(26841) LOG: Done calling Route 2
Anyone know what my problem is here?
Thanks.