Hello,
I have two issues related to dialog module state keeping/logging that
caught my attention today:
1.) The module considers requests routed in early (but not yet
confirmed) dialogs to be bogus, as can be seen from a few lines of state
machine code in dlg_hash.c:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=modules_…
In the logs, it shows up something like this:
CRITICAL:dialog:log_next_state_dlg: bogus event 8 in state 2 for dlg
0x7f9f237ffff8 [2901:1286709577] with clid '<Call-ID>' and tags '<caller
tag>' '<callee tag, always empty>'
I believe that such in-early-dialog requests should not trigger
bogus-event log messages as they look fine from a standard's point of view.
2.) Various race conditions may generate log messages such as following
when DID mode is enabled:
WARNING:dialog:dlg_onroute: unable to find dialog for <REQUEST TYPE>
with route param 'faf.f69e7b4' [4015:79161711]
Observed REQUEST TYPEs at our site were "BYE" when BYE requests were
transmitted by both caller and callee at roughly the same time due to
user behavior; and "PRACK" when a PRACK was received by the UAS after a
486 (Busy Here)/ACK pair of messages was already exchanged between the
UAS and the proxy. In both cases, the associated dialog had already been
destroyed when another request was received.
My suggestion is to lower the log level for this kind of message because
it can simply happen too often but doesn't affect proper dialog handling.
Feedback?
Cheers,
--Timo
Dear ALL,
I have configured Kamailio 1.5 from svn with radius (radiusclient-ng)
support for acc.
All seem to work okay, except one thing.
>From time to time I am receiving following radius error (debug from
radius server on different ip)
Fri Jul 9 12:22:58 2010 : Error: rlm_sql (sql): Couldn't update SQL
accounting STOP record - Column 'AcctStartTime' cannot be null
Fri Jul 9 12:23:08 2010 : Error: rlm_sql (sql): Couldn't update SQL
accounting STOP record - Column 'AcctStartTime' cannot be null
Fri Jul 9 12:23:18 2010 : Error: rlm_sql (sql): Couldn't update SQL
accounting STOP record - Column 'AcctStartTime' cannot be null
A few lines from radiusclient.conf
# time to wait for a reply from the RADIUS server
radius_timeout 10
# resend request this many times before trying the next server
radius_retries 3
After closer look at this, it turned out that there is one START
request and four STOPs with the same CallID (Acct-Session-Id).
START and first STOP are treated as a pair, the next ones are odd.
Taking about sip i do have following scenario:
MGW(1)---Kamailio(2)---SIP_Provider(SER based)(3)
The call flow that generate aforemtioned errors looks like below
(debug on kamailio)
1 - (2) is receiving INVITE from (3)
2 - (2) is sending Giving a Try to (3)
3 - (2) is sending INVITE to (1)
4 - (2) is receiving Trying from (1)
5 - (2) is receiving Ringing from (1)
6 - (2) is receiving OK from (1)
7 - (2) is sending OK to (3)
8 - (2) is receiving ACK from (3)
9 - (2) is sending ACK to (1)
---- call----
10 - (2) is receiving BYE from (1) (tearing down from (1) side, STOP
radius request is generated)
11 - (2) is sending BYE to (3)
12 - (2) is receiving BYE from (3) (second STOP is generated,
radius_retries 3, radius_timeout 10)
13 - (2) is sending BYE to (1)
14 - (2) is receiving OK from (1)
15 - (2) is sending OK to (3)
16 - (2) is receiving OK to (3)
17 - (2) is sending OK to (1)
For sure there is sth wrong with SIP_Provider but how Kamilio should
behave is such situation.
Should second BYE (p.12) with the same callid as in p.10 be continued with p.13?
Thanks in advance,
Maciej.
Hey guys,
I'm using Kamailio 1.5.3-notls and need to apply some re operation to an AVP.
The problem I have is that my re expression is held in an AVP, so the
parser is not recognizing it :(
Would it be very hard to change that behavior? Any ideas/tips to implement?
This code
route[16]
{
...
avp_subst("$avp(s:ANI)/$avp(s:ANIegress)", "$avp(s:carrierAniRegex)");
...
}
returns this error
ERROR:avpops:fixup_subst: avpops: bad subst re $avp(s:carrierAniRegex)
This code
route[16]
{
...
avp_subst("$avp(s:ANI)/$avp(s:ANIegress)", $avp(s:carrierAniRegex));
...
}
returns this error
CRITICAL:core:yyerror: parse error in config file, line 751, column
51-74: syntax error
CRITICAL:core:yyerror: parse error in config file, line 751, column
74-75: bad arguments
Do you think some other way I can achieve this? Be aware I'm using
SQLops to get realtime the value of the expression...
Thanks in advance!
Uriel
Hey,
I have setup two Kamailio servers, my INVITEs will go through server A,
which forwards to server B which forwards it back to server A.
This is intential, the trouble is when the packet comes back through
server B, nothing has changed so TM sees it as a retransmission.
The difference between the first time through and the second time is the
port 5061 instead of 5060.
How can I tell tm that these are not the same transactions ? Would it be
easier to have a seperate instance of Kamailio for 5061 ?
Thanks,
David
Hi all,
I'm using kamailio 1.5 with TLS module.
I need to make ENUM query and get NAPTR record.
>From NAPTR lookup, I'd like to relay my SIP Invite with tls protocol.
How can I tell Kamailio to use TLS protocol ( instead of udp) after NAPTR lookup ?
I've try to set :
dns_tls_pref=1
dns_udp_pref=2
dns_tcp_pref=3
in the general section of kamailio.cfg, but I get a parse error.
Regards,
Daniel
Hi klaus,
Suppose I can't access to NAPTR settings.
I need to manage SIP URI, so , If I right understand, the only way to
use TLS protocol in kamailio 1.5 is to append ";transport=tls" in R-URI
before relay.
In other words I need to rewrite R-URI:
$ru = $ru + ";transport=tls" ;
# and the t_relay
t_relay() ;
something like that?
Regards,
Daniel
Il 08/07/2010 18.45, Matteo Campana ha scritto:
>
>
> -------- Messaggio originale --------
> Oggetto: Re: [SR-Users] Kamailio and NAPTR lookup with TLS
> Data: Thu, 08 Jul 2010 18:44:27 +0200
> Mittente: Klaus Darilion <klaus.mailinglists(a)pernau.at>
> A: Daniel-Constantin Mierla <miconda(a)gmail.com>
> CC: matteo.campana(a)klarya.it, sr-users(a)lists.sip-router.org
>
>
>
> Am 08.07.2010 18:10, schrieb Daniel-Constantin Mierla:
> > Hello,
> >
> > On 7/8/10 5:59 PM, Matteo Campana wrote:
> >>
> >> Hi all,
> >> I'm using kamailio 1.5 with TLS module.
> >> I need to make ENUM query and get NAPTR record.
> >> > From NAPTR lookup, I'd like to relay my SIP Invite with tls protocol.
> >>
> >> How can I tell Kamailio to use TLS protocol ( instead of udp) after
> >> NAPTR lookup ?
> >>
> >> I've try to set :
> >>
> >> dns_tls_pref=1
> >> dns_udp_pref=2
> >> dns_tcp_pref=3
> >>
> >> in the general section of kamailio.cfg, but I get a parse error.
> >>
> > these parameters were introduced in kamailio with version 3.0.
> >
> > If you need TLS then it is recommended to use 3.0 anyhow, it is a far
> > better implementation. That will make the life easier to migrate to
> > upcoming 3.1 that will bring asynchronous TLS.
> >
> > No matter what you have in R-URI, you can force TLS via setting outbound
> > proxy address to be a TLS uri:
> >
> > $du = "sip:__ip_or_host__;transport=tls";
> > t_relay();
>
> IIRC we do have NAPTR support in Kamailio 1.5 - don't we?
>
> Then I think it should work when putting a domain into $du and makeing
> sure that there is no transport parameter, no port, and NAPTR TLS record
> has highest priority.
>
> regards
> klaus
>
> >
> > The IP or host you can take from R-URI without any problem via PV $rd.
> > Other option is to use function from tm - t_relay_to_tls():
> >
> > http://kamailio.org/docs/modules/stable/modules/tm.html#t_relay_to_udp
> >
> > Cheers,
> > Daniel
> >
>
>
Hello all,
I need to work on particular ACK within the configuration of SER 0.9.9.
Firstly, I was thinking to make a difference using the CSeq field, but
it seems that the CSeq on ACK contains the ACK himself. Probably because
ACK is a request and not a response.
Would it be possible to proceed in another way?
Regards,
Adrien .L
Hi, I new with kamailio, I've been able to integrate kamailio 3.02
with asterisk 1.6. The only issue I'm having is if I have to restart
asterisk( for some config update) I loose all the sip registration in
asterisk, is there any way of fixing this?
Hi,
When I build sip-router (from master) with kamailio flavour using:
make config mode=debug FLAVOUR=kamailio; make all; make install
I'm getting this error when trying to start kamailio:
ERROR: <core> [sr_module.c:396]: ERROR: load_module: could not open module
</usr/local/lib/kamailio/modules_k/tmx.so>:
/usr/local/lib/kamailio/modules_k/tmx.so: undefined symbol: _tm_table
On the other hand, if I use:
make FLAVOUR=kamailio cfg; make all; make install
kamailio starts correctly.
I've tried also:
make mode=debug FLAVOUR=kamailio cfg
with no luck either.
What's the problem? How should I compile kamailio with debugging so it
starts correctly?
Thank you in advance.
Regards,
Santi
