Ser doesn't appear to be passing the Caller-id to the ata at auth or I
am doing something wrong. Can anyone point me in the right direction?
Thanks,
Stephen
I have the following entry in my freeradius users file.
test(a)voip2.test.net Auth-Type := Digest, User-Password == "test"
Reply-Message = "Hello, test with digest", Sip-Rpid =
"8472222222"
When I run a radclient test I get the correct info..
radclient -f digest.test 219.242.10.153:1812 auth testing
Received response ID 134, code 2, length = 57
Reply-Message = "Hello, test with digest"
Sip-Rpid = "8472222222"
This is the output from ngrep port 5060
U 216.222.234.113:5060 -> 219.242.10.153:5060
REGISTER sip:voip2.test.net SIP/2.0..Via: SIP/2.0/UDP
216.222.234.113:5060..
From: sip:test@voip2.test.net;tag=277486986..To:
sip:test@voip2.test.net..Cal
l-ID: 2687235586@216.222.234.113..CSeq: 2 REGISTER..Contact:
<sip:test@216.
222.234.113:5060;transport=udp>;expires=120..User-Agent: Cisco ATA 186
v2.
16.2 ata18x (030909a)..Authorization: Digest
username="test",realm="voip2.test.net",nonce="3f7c49b40e81572eff05bdf50c
867a85bbb0da3c",uri="sip:voip2.test
.net",response="1684410c130d6faa9a3c573365f36ab6"..Content-Length:
0....
#
U 219.242.10.153:5060 -> 216.222.234.113:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP
216.222.234.113:5060;rport=5060..From: sip
:test@voip2.test.net;tag=277486986..To:
sip:test@voip2.test.net;tag=b27e1a1d3
3761e85846fc98f5f3a7e58.13c2..Call-ID:
2687235586@216.222.234.113..CSeq: 2
REGISTER..Contact:
<sip:test@216.222.234.113:5060;transport=udp>;q=0.00;exp
ires=120..Server: Sip EXpress router (0.8.12dev-t16
(i386/linux))..Content-
Length: 0..Warning: 392 219.242.10.153:5060 "Noisy feedback tells:
pid=121
59 req_src_ip=216.222.234.113 req_src_port=5060
in_uri=sip:voip2.test.net ou
t_uri=sip:voip2.test.net via_cnt==1"....
Daniel,
Yes the windows messenger (5.0) that i am using supports SIP. So now how
do i go about it. I did configure the messenger to include SIP address. I
want to check this user record on the server. How do i do that? Should i
send my ser.cfg to you just to check my settings. Please let me know if it
is feasible.
Thanks,
Annie
Thanks Daniel,
I want to allow any user in the SIP domain to be able to register since i
have control over that. What i cant figure out is when i log into Windows
Messenger( which right now is not supporting SIP temporarily ) with my SIP
user name shouldnt i be registered in the SIP server? How do i check this
cause i dont seem to find any record of any user. Since the registrar
modules are loaded by default i assume that the server is working as a
registrar server too. Am i wrong or am i missing some info.
Please guide.
Thanks for the support.
Annie
Maxim,
I am still having a problem accessing this site. Is it up yet? Can
you send me the latest tarball please?
-Josh
>
>
Dovid wrote:
> Does anyone know of a working site to download rtpproxy?
> https://demo.portaone.com/~sobomax/PortaSIP is not responding.
> (Does anyone have a copy to send me?)
Yes, I know, we are moving our server to the new location, so that it is
off-line now. It should be available later today, if not, drop me a note
I'll mail you a tarball.
-Maxim
Hello,
I got the debug messages -- if the mail is too large, send it to
serteam(a)iptel.org instead of serusers(a)lists.iptel.org -- this one has size
limitation.
The INVITE is processed correctly, the Request-URI is replaced with the
contact address of the destination user (2222) and then the request is
forwarded but the client does not react.
What kind of SIP client do you use? Both clients are registered from
same IP address and port (202.133.79.3:5060).
}Daniel
On 10/2/2003 11:50 AM, John Foster wrote:
> Previous mail stuck for moderator, due to larger size of mail..
>
>
> Hi All n Daniel,
>
> Here are network Dumps..
> Sorry for too lengthy dumps:) These include first portion of starting
> up ser -d -d -E, then both soft phone logged in, lastly there are
> traces of call initiating attempts from user 12345(a)cooking.com.pk
> <mailto:12345@cooking.com.pk> to user 22222(a)cooking.com.pk
> <mailto:22222@cooking.com.pk>
>
>
> Regards,
> JF
>
Hi All,
A new user here,
I m using 0.8.11, with default conf file shipped with package.without auth. While my softphone registers with the proxy, it gets listed in "serctl ul show", but another user while joins through same proxy with all other params same except username. The two cannt dial eachother.their addresses URIs are 12345(a)xxx.yyy.com and 22222(a)xxx.yyy.com, while caller,callee and proxy are in same IP subnet. They gett Timeout message.
Can anyone give any idea?
Thanks in Adv.
JF
---------------------------------
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Thanks Daniel,
I want to allow any user in the SIP domain to be able to register
since i have control over that. What i cant figure out is when i log into
Windows Messenger( which right now is not supporting SIP temporarily ) with
my SIP user name shouldnt i be registered in the SIP server? How do i check
this cause i dont seem to find any record of any user. Since the registrar
modules are loaded by default i assume that the server is working as a
registrar server too. Am i wrong or am i missing some info.
Please guide.
Thanks for the support.
Annie
Hi,
I'm trying to change the destination set of an request in a failure
route using exec_dset(). My routing looks like
route {
...
t_on_failure(1);
...
}
failure_route[1] {
revert_uri();
exec_dset("find-voicemail.pl");
t_relay();
break;
}
unfortunately, that does not seem to work, i get the following error
from SER when a request hits the failure_route[]:
Oct 1 14:33:17 graham /usr/sbin/ser[4814]: ERROR: t_forward_nonack: no
branched for fwding
Oct 1 14:33:17 graham /usr/sbin/ser[4814]: ERROR: w_t_relay (failure
mode): forwarding failed
Simply adding a static destination by append_branch() works fine, but
that's only the solution for an announcement... I'd need something like
exec_append_branch() ...
So, is there another way to dynamically add branches? Any hints
appreciated!
Alexander Mayrhofer
nic.at
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Maxim,
I am still having a problem accessing this site. Is it up yet? Can
you send me the latest tarball please?
-Josh
>
>
Dovid wrote:
> Does anyone know of a working site to download rtpproxy?
> https://demo.portaone.com/~sobomax/PortaSIP is not responding.
> (Does anyone have a copy to send me?)
Yes, I know, we are moving our server to the new location, so that it is
off-line now. It should be available later today, if not, drop me a note
I'll mail you a tarball.
-Maxim