Hi All,
What does this error mean?
Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: Maxfwd module-
initializing
Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: ERROR: auth_radius: can't
get code for the Sip-Session attribute value
Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: init_mod(): Error while
initializing module auth_radius
I'm using ser 0.9.5 and radiusclient-ng 0.5.2.
Thank You
Regards,
Nhadie
Hi All,
What does this error mean?
Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: Maxfwd module-
initializing
Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: ERROR: auth_radius: can't
get code for the Sip-Session attribute value
Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: init_mod(): Error while
initializing module auth_radius
I'm using ser 0.9.5 and radiusclient-ng 0.5.2.
Thank You
Regards,
Nhadie
Hi All,
What does this error mean?
Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: Maxfwd module-
initializing
Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: ERROR: auth_radius: can't
get code for the Sip-Session attribute value
Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: init_mod(): Error while
initializing module auth_radius
I'm using ser 0.9.5 and radiusclient-ng 0.5.2.
Thank You
Regards,
Nhadie
Hello,
This time I've got a question :)
I think it will be useful for everybody.
I have discussion with SIP device manufacturer about expiration time of
call attempt. When call is made from UA it sends INVITE with "Expires:
65". It is controlled by device's timer setting "ring_time_limit". This
timer can be set to anything between 10 and 600 seconds, we set it to 65
seconds. Fine.
However at the same time it starts a special timer, say
"max_response_time_invite". It can be set to anything between 8 and 20
seconds. We set it to 20 sec. Fine so far.
CASE 1: Imagine we call from UA to PSTN number, using one of VoIP->PSTN
providers. We have more than one provider, so we setup SER with
"fr_timer" value of, let's say 25 seconds and "fr_inv_timer" of 60. We
prepare failure routes in case when those timers hits in PSTN call. When
"fr_timer" hits, we simply reroute to another gateway. Great.
SER forwards INVITE from UA to remote PSTN gateway and at the same time
sends back "100 Trying" to UA. Fine.
Imagine now that it is busy-hour and it takes PSTN provider 21 seconds
to send back "180 Ringing" (or "183 Session Progress") and after 8
seconds more remote callee picks up ("200 OK").
Unfortunately device wil send CANCEL to SER, because it hasn't received
"180 Ringing" or "183 Session Progress" within the
"max_response_time_invite" setting of 20 seconds. It only received "100
Trying" at the beginning.
This ruined any failure route attempts from SER, as call failed.
CASE 2: Imagine now a second call. This time PSTN provider send "183
Session Progress" after 14 seconds and remote callee picked up after
further 15 seconds. This time everything went good. UA did not hit
"max_response_time_invite" timer - it waited patiently for whole 29
seconds for remote callee to pick up.
I think that when UA receives provisional response (like "100 Trying")
from SER it should stop timer "max_response_time_invite" and use
'ring_time_limit'. What device does now, it uses:
* "max_response_time_invite" when "100 Trying" received
* "ring_time_limit" when "180 Ringing" or "183 Session Progress" received
I haven't really found explanation of this in RFC3261. When you look at
state machine on page 128, it is not explicitly said which timer
controls behaviour of UA when in "Proceeding" state ("Proceeding" is
entered oin receipt of ANY provisional response).
QUESTION 1: what timer controls behaviour of UA when in "Proceeding"
state? Especially regarding timeout. Is this still the 'timer B' started
at the beginning of INVITE transaction? If it is 'timer B', why UA is
CANCELing transaction after 20 seconds, if it informed by "Expires: 65"
field in INVITE that it set 'timer B' to 65 seconds.
QUESTION 2: What should I propose to manufacturer for maximum value for
"max_response_time_invite"? Something like 120 seconds ?
--
Thanks,
Arek Bekiersz
Hi
I'd like to use PA, XCAP with eyeBeam.
Are you talking about the "ser-0.10.99-dev30-tm-timers-pa-3", I've
downloaded the tar file and tried "make install".
It seems like presence works fine, XCAP is also working(except RLS)...
About the RLS, do I need to create a directory in
"<xcap-root>/rls-services/global/index" where all the XML (for RLS)
files are stored?
How can I configure the eyeBeam accordingly? By default it's configured
as "subscribe to contact list : $username$-list@$domain$".
It seems like SER does not consider it as a list, it only considers it
just another URI. Did I do something wrong? Or do I need to change
anything in SER?
Regards,
Dennis
-----Original Message-----
From: Jan Janak [mailto:jan@iptel.org]
Sent: Tuesday, January 24, 2006 11:11 PM
To: Yeung OnTai-q16645
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] re : presence module with ser
On 18-01-2006 09:07, Yeung OnTai-q16645 wrote:
> Hi all,
>
> I need to build SER with presence and XCAP. Does anyone know which
> version of SER contain a relatively "stable" version of "pa" and
"rls".
> I've tried 0.9.6 and the cvs stable build, but I ran into different
> problems(error in loading modules, compling errors, etc) in making the
> source code.
>
> Can anyone point me to the right direction please?
Try the version from http://ftp.iptel.org/pub/ser/presence
That one should be relatively easy to build. Note that the XCAP server
is not included.
Jan.
>
> Thank you very much!
>
> Regards,
> Dennis
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
OK, after some investigations here are my findings. I post them here so if
some one get the same error he might get hints.
It seems if the the other end of the call has his soft phone closed, this
error is returned. I tried to increase the max size to 4096 and now I get
the "Too many hops" error. Which means, the message is forwarded way too
much time, beeing increased over the 2048 limit.
Now, why do I get a route loop when a user is offline...I suspect this
"...if (uri==myself) {..." of beeing the culprit. I added various alias to
the proxy ip adress, to the proxy name and even the domain name bu nothing
worked out.
Any ideas ?
Dear SER users,
I cannot get any information on the column definition of the 'location'
table. However, all I really need is the definition of the 'flags' column. I
know if you set flags to 128 you insert a permanent entry into the location
table, but I cannot figure out what the rest of the flags are for. Can
anyone help?
I have searched:
- seruser archives
- location module readme file
- onsip website
Leo Papadopoulos
Web site: www.telecomCTO.com
Okay... I'll admit, this isn't the most recent PA module, but I thought
someone might have a hint nonetheless.
I'm running SER 0.9.6. I do NOT wish to upgrade to the somewhat haphazard
presence snapshot as it's not entirely in stable/usable condition yet. I was,
however, under the impression that simple PUBLISH/SUBSCRIBE methods would work
okay on this version of the module. Am I mistaken?
Usual stuff in the ser.cfg
if(method=="SUBSCRIBE")
{
if(!t_newtran())
{
log(1, "SUBSCRIBE newtran error\n");
sl_reply_error();
};
handle_subscription("registrar");
break;
};
if(method=="PUBLISH")
{
if(!t_newtran())
{
log(1, "PUBLISH newtran error\n");
sl_reply_error();
};
handle_publish("registrar");
break;
};
When my Snom phone sends a publish, it looks like this:
U 66.112.115.10:2057 -> 66.112.115.34:5060
PUBLISH sip:11019552444@my.domain.com:5060 SIP/2.0.
Via: SIP/2.0/UDP 66.112.115.10:2057;branch=z9hG4bK-nc00a965o8y5;rport.
From: "Death" <sip:11019552444@my.domain.com:5060>;tag=o6bezw24ge.
To: "Death" <sip:11019552444@my.domain.com:5060>.
Call-ID: 3c267063afc8-19wbm3ssr4y3@snom190.
CSeq: 1 PUBLISH.
Max-Forwards: 70.
Event: proxy-config.
Content-Type: application/text.
Content-Length: 0.
.
It sends this (with slightly different From: tags) 9 times and then ser
coredumps without a word to the log.
Any ideas? Give up presence for now and just tell my users to get over it? :)
N.
Hi!
Reading the TLS docs I understand that using the tls_domain section, I
can specify different certificates and CAs which will be used according
to the socket on which the incoming TLS request is received.
But is there also a way to use the "TLS domains" also for outgoing TLS
connections? If yes, how do I trigger using a certain "TLS domain" for
outgoing TLS connections?
regards
klaus