Hi,
I saw those messages ... but at that time i was not onto testing the
tls code, sorry,
so i had no experience to report back.
I managed to compile it using ser 0.9.x (sorry ... i don't remember the
minor version by
heart).With this i mean that it compiles and runs, though i am not
using any of the tls
domains. I use only one, with the default certificates and so on.
And definitely ... i am going to try the t_relay_to_tls ... i
completely forgot about that
function :D
Regards,
Cesc
>>> Juha Heinanen <jh(a)tutpro.com> 03/16/05 08:47PM >>>
Cesc Santasusana writes:
> There have been no reports back, i guess either people are not very
> interested (i dont know why, i am very much) ...
people are interested. i saw a message on the list where someone had
tried to compile the code and had failed. then he asked about which
version the code should run, i.e., 0.9.0 or unstable, but i didn't see
any reply on the list.
> I tried to install it, and compiled without problems. I was hoping
to
> use minisip against it,
which version of ser did you use?
> but it would be too much beta-testing. It seems it deals ok with
> connections and so on
> (i tried connecting with a browser via https to the port where ser
> listens for tls ... it worked,
> but of course the parsing of the message failed ... as it was to be
> expected).
> I am trying to develop an injector, to use in conjunction with
sipsack
> or the like ... keep
> u posted.
perhaps you could try between two proxies using t_relay_to_tls?
-- juha
Unclassified
Hello list,
Want to add that I never got this error message (error: 400; check if you
use alises in SER) when I installed 0.9.0 stable and 0.8.14, it only happens
when I installed from the unstable version. Could someone help me out
please?
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Hi all,
I am looking for a solution for a situation wher I have multiple
asterisk systems available, each supposed to handle a different set of
dialed prefixes (like areacodes). I don't want to describe each fo
these in ser.cfg, what I am wodnering is can LCR module be used for
this purpose - to provide a dynamic routing decision for the different
prefixes?
Thank you.
Hi.
I want to sound three telephones at a time.
And, the call of other telephones stops, too, when one in that receives the telephone.
I wrote ser.cfg like this.
if (uri=~"^sip:1000@.*"){
rewriteuri("sip:Aphone@foo.bar");
append_branch("sip:Bphone@foo.bar");
append_branch("sip:Cphone@foo.bar");
t_relay();
break;
};
Then communication is done, but it is cut off immediately.
The error log of that time is here.
ERROR: parse_uri: uri too short: <sip:> (4)
get_username(): Error while parsing R-URI
insert_RR(): Error while extracting username
record_route(): Error while inserting Record-Route line
ERROR: parse_uri: uri too short: <sip:> (4)
ERROR: parse_sip_msg_uri: bad uri <sip:>
WARNING: do_action:error in expression
ERROR: parse_uri: uri too short: <sip:> (4)
ERROR: parse_sip_msg_uri: bad uri <sip:>
ERROR: new_t: uri invalid
ERROR: t_newtran: new_t failed
ERROR: sl_reply_error used: Regretfuly, we were not able to process the URI (479/SL)
Is this method wrong?
If there was a good idea, please teach it.
Thanks in advance.
Jun
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Hi there,
I'm using ser and asterisk together. Asterisk for voice mail etc and ser for registration of the users
usig database. I can restrict forwarding calls from another sip proxy to ser (using proxy_authorize) but how can I restrict access to asterisk ... Now everyone can forward calls to my asterisk and can place pstn calls.
Thanks in advance,
Steve
Can anyone help with some meaningful checks of SER using sipsak and
Nagios? My SER box is backended with RADIUS, and I can do a "spisak -s
blah@server" and get a 200 for a registered UA, but I'd really like to
know what other generic SIP tests people are performing using sipsak and
Nagios together (perhaps a sample from your checkcommands.cfg?)
I understand that sipsak supports Nagios' return codes, which is cool, I
just need to understand how to make best use of it. Documentaion is
scarce, and extensive searching of the lists have proved largely fruitless.
Thanks in advance.
Hi,
I have in my ser.cfg:
modparam("tm","fr_inv_timer",40)
set as a timer so that INVITEs coming from PSTN callers go to
voicemail after 40 seconds. This works great for inbound called,
however it seems to be a limitation for outbound calls. If the
outbound INVITE doesn't receive an ACK in that time period, the call
gets cancelled and drops to fast busy.
Is there a way around this? Can I set different timers for inbound and
outbound calls? 40 seconds seems to be a good amount time so that
most outbound calls get answered by voicemail or actual people, but
occastionally users will still get a fast busy.
Any advice is greatly appreciated.
Dan
