Hi,
Not sure if someone has asked this before. Is there any way to route all SIP
messages through SER without using record_route()?
Jaime
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Hi,
I have successfully registered the Windows Messenger (4.7) to SER. I was also able to make SIP calls to PSTN by entering the number by selecting "Start a Voice Conversation" -> "Other" -> "Enter e-mail address". But how to enter the number directly from the dialer? It seems that the dialer doesn't work (no response) so I have no way to enter DTMF for IVR applications.
Anyone has any idea? Thanks.
Reagards,
Bo
Hi,
I was reading about the ATA186's ability to detect that it is behind a NAT.
It requires that:
"For the Cisco ATA to automatically detect its presence behind a NAT, the
SIP proxy server
or remote user agent server must include the "received=" parameter in the
Via header in the
responses to the Cisco ATA if the proxy detects that the source address and
port do not match
those in the Via header."
Does SER support it?
Thanks,
Ricardo
Hi,
I am trying to use t_uac_dlg to send a new message from a module. It does
not send the message, and I get the following in my log:
Mar 21 14:50:54 d254196 /usr/local/sbin/ser[3111]: ERROR: mk_proxy: could
not resolve hostname: "cs.stir.ac.uk"Mar 21 14:50:54 d254196 /usr/local/sbin/ser[3111]: ERROR: t_relay: bad
host name in URI <sip:srm@cs.stir.ac.uk>Mar 21 14:50:54 d254196 /usr/local/sbin/ser[3111]: ERROR: uri2sock: Can't
create a dst proxyMar 21 14:50:54 d254196 /usr/local/sbin/ser[3111]: ERROR: t_uac_dlg: no
socket found
The code in my module is as follows, where all the ps_ things are str*
variables, with values as indicated by comments:
tmb.t_uac_dlg( // send request
ps_req_type, // INVITE
NULL,
PROTO_UDP ,
ps_req_uri, // sip:srm@cs.stir.ac.uk
ps_req_to, // srm(a)cs.stir.ac.uk
ps_req_from, // mrs(a)cs.stir.ac.uk
NULL,
NULL,
NULL,
NULL,
ps_req_headers, // NULL
ps_req_body, // NULL
NULL, // transaction_cb callback param //
optional NULL // void* callback pointer // optional
);
Any hints? I might have misunderstood what is expected from a request uri.
If so, an example would be very welcome.
thanks
Stephan
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I am trying to bring up the anthentication module. Here are the list of what I have done.
1. Create MySQL tables and modify the permissions (SER could connect MySQL successfully)
2. Modify SER configuration (See below)
loadmodule "//usr/lib/ser/modules/mysql.so"
loadmodule "//usr/lib/ser/modules/auth.so"
modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser")
modparam("usrloc", "db_mode", 2)
modparam("auth", "db_url", "sql://ser:heslo@localhost/ser")
modparam("auth", "user_column", "user_id")
modparam("auth", "calculate_ha1", yes)
modparam("auth", "password_column", "password")
if (uri=~"sip\.abcd\.com") {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("sip.abcd.com", "subscriber")) {
www_challenge("sip.abcd.com", "0");
break;
};
save("location");
break;
};
After all these changes, SER could start without problem. But it seems that the Auth module doesn't take effect. For whatever username/password (whether it's included in the subscriber table or not), SER just allow the registration.
What maybe the problem? Is there anything wrong with the configuration?
Thanks.
Hi,
I tested auth_radius module with freeradius and it works perfecly
But now i would like test it with Microsoft IAS.
Is it possible ?
Even if i had Attribute in Dictionary
I have this error on IAS Event :
A malformed request was received from client Sip. The data is the packet.
Thank you for your help
Yours faith fully
I know I can do this, but I can't figure out how.
How can I put a button on my web site that says 'call me'.
Given two SIP accounts, assuming they are registered, can I
connect the two???
---greg