Hi
Can anyone point to a good example of Kamailio performing the function of Re-direct server.
We would like to forward messages onto another domain for authentication.
Hi guys
I have a multihomed Kamailio proxy sitting between two B2BUAs on
separate networks and record-routing all dialogs.
The problem I have is that when one of these devices receives a 200
OK, it does not populate the RURI of the ACK correctly. Instead of
taking it from the Contact header on the 200 OK, it uses the user part
from the Contact header and sets the domain to the proxy IP. It then
also populates the Route headers.
The ACKs are below and the IPs are:
10.152.1.92:5060: UAC on outside
10.200.70.100:5060: proxy outside interface
192.168.242.100: proxy inside interface
192.168.242.102: UAS on inside
This the ACK message going into and coming out of the proxy.
# U 10.152.1.92:5060 -> 10.200.70.100:5060
ACK sip:natted_ua*9197**192.168.242.102*5080*udp@10.200.70.100 SIP/2.0.
Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK-2280-1-5.
Route: <sip:10.200.70.100;r2=on;lr=on;nat=yes>,<sip:192.168.242.100;r2=on;lr=on;nat=yes>.
From: sipp <sip:sipp@127.0.1.1:5060>;tag=2280SIPpTag001.
To: sut <sip:9197@10.200.70.100:5060>;tag=p0FaeQ1QUS9ae.
Call-ID: 1-2280(a)127.0.1.1.
CSeq: 1 ACK.
Contact: sip:sipp@127.0.1.1:5060.
Max-Forwards: 70.
Subject: Performance Test.
Content-Length: 0.
.
# U 10.200.70.100:5060 -> 192.168.242.100:5060
ACK sip:192.168.242.100;r2=on;lr=on;nat=yes SIP/2.0.
Via: SIP/2.0/UDP 10.200.70.100;branch=z9hG4bKcydzigwkX.
Via: SIP/2.0/UDP
127.0.1.1:5060;rport=5060;received=10.152.1.92;branch=z9hG4bK-2280-1-5.
Route: .
From: sipp <sip:sipp@127.0.1.1:5060>;tag=2280SIPpTag001.
To: sut <sip:9197@10.200.70.100:5060>;tag=p0FaeQ1QUS9ae.
Call-ID: 1-2280(a)127.0.1.1.
CSeq: 1 ACK.
Contact: <sip:sipp@127.0.1.1:5060;alias=10.152.1.92~5060~1>.
Max-Forwards: 69.
Subject: Performance Test.
Content-Length: 0.
So I am trying to understand why it is trying to relay to itself (I
have mhomed=1) and why it is rewriting the RURI as if it is a strict
router.
Currently decode_contact() is disabled but enabling doesn't seem to
help. Unless there is a very specific place where it belongs.
Any advice would be hugely appreciated. I can always paste logs /
configs / traces.
Richard
--
Richard Brady
M: +44 (0)7771 623 348
T: +44 (0)20 8144 8160
E: rnbrady(a)gmail.com
Hello,
I'm using uac_req_send () to send a request generated in Kamailio (eg
Example 1.21. UAC module)
$ uac_req (method) = "OPTIONS";
$ uac_req (ruri) = "sip: kamailio.org";
$ uac_req (furi) = "sip: kamailio.org";
$ uac_req (turi) = "sip: kamailio.org";
uac_req_send ();
However, I need to capture the responses of this request with failure_route
TM module.
Is there a way to capture these responses when I'm using uac_req_send()
function?
I appreciate your comments.
Freddy
Hello,
I am thinking of having our next IRC devel meeting soon, to plan the
next major release and review current stable releases and the
environment around the project (e.g., if you can add anything else to
the project to make life easier for developers and users).
I'm proposing next week, Thursday, at 15:00GMT, on Freenode IRC server,
channel #sip-router.
Anyone able to join? If it is not a convenient date for you, just
propose alternatives and we will select the one that meets the
constraints of the most developers and users willing to participate.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Hello,
I've been searching the internet to find an explanation on how SIP transfer works using Re-INVITE and/or UPDATE, but I can't seem to find a good source.
>From what I understand(and this is the way we do it), the following happens:
Bob=Caller
Alice=Called
John=Transfer party
1) Bob calls Alice. The usual INVITE,Trying,200 OK, ACK.
2) Alice transfers the call to John using Re-INVITE.
a. Alice calls John. The usual INVITE,Trying,200 OK, ACK.
b. Alice Re-INVITEs Bob using INVITE with adjusted SDP.
3) Bob is connected to John through Alice in some magical way. I'm guessing because the SDP has been changed and for some reason the RTP stream flows between Bob and John through Alice?
Is this correct? If not, perhaps someone could explain it to me from scratch.
Maybe useful to know that we are using Cisco equipment for call handling (VXML and TCL scripts).
Thanks,
Grant
Hi All
Can anyone point to documentation tutorial on howto forward statefully a REGISTER message to another SIP Server. I have looked at http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb, However I'm getting errors with the uac_req commands even though I have loaded the uac.so module. .
I've been working on a Shared Call Appearances module for the past several months. It implements the Broadsoft SCA feature as laid out in Broadworks SIP Access Side Extensions Interface Specifications document. (Another implementation of the same feature was added to Freeswitch a few years ago, but we didn't want to use Freeswitch.)
We've been testing and improving the module over the last month, and I think it's ready to share. The module is available at GitHub here:
<https://github.com/fitterhappier/sca>
To date, we've only tested the module with Polycom handsets.
I'm happy to answer questions. I hope this will prove useful for others.
Best,
andrew
IPTEL
We needhelp or cost for the development a script for kamailio which can
do the following:
Register kamailio to another gateway using UAC modules.
route all traffic coming towards a private ip 10.x.x.x
We will use a postgres database
We have installed kamailio 3.1 with postgres support and UAC modules
Rubén Mejía
Telefonia Cabletamps
hello, guys:i follow the linkhttp://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-a… can registered users, kamctl online shows those status. but i can not make calls. this is debug message when i make a call from 102 to SIP 103--------------------------------------------------------------- 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=486 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=3 n=return 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=489 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=636 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=601 a=25 n=has_totag 8(1286) DEBUG: siputils [checks.c:103]: no totag 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=501 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=494 a=26 n=is_method 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=496 a=25 n=t_check_trans 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [tm.c:1015]: lookup_original: t_lookupOriginalT returned: 0xb4d36320 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=497 a=25 n=t_relay 8(1286) DEBUG: tm [t_lookup.c:1379]: DEBUG: t_newtran: msg id=29 , global msg id=28 , T on entrance=(nil) 8(1286) DEBUG: tm [t_lookup.c:527]: t_lookup_request: start searching: hash=60763, isACK=0 8(1286) DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed 8(1286) DEBUG: tm [t_lookup.c:709]: DEBUG: t_lookup_request: no transaction found 8(1286) DEBUG: tm [t_hooks.c:374]: DBG: trans=0xb4d3a698, callback type 1, id 0 entered 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [t_fwd.c:1243]: DEBUG: e2e_cancel: e2e cancel proceeding 8(1286) DEBUG: <core> [msg_translator.c:206]: check_via_address(192.168.1.103, 192.168.1.103, 0) 8(1286) DEBUG: <core> [mem/shm_mem.c:111]: WARNING:vqm_resize: resize(0) called 8(1286) DEBUG: tm [t_reply.c:1543]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 8(1286) DEBUG: tm [t_reply.c:703]: DEBUG: reply sent out. buf=0xb721ee38: SIP/2.0 200 cancelin..., shmem=0xb4d38c08: SIP/2.0 200 cancelin 8(1286) DEBUG: tm [t_reply.c:713]: DEBUG: _reply_light: finished 8(1286) DEBUG: tm [t_funcs.c:388]: SER: new transaction fwd'ed 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=3 n=exit 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list=================================anyone can give a hint for that? thanks!
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording device, VOIP gateway.
website: www.hiastar.com
> From: sr-users-request(a)lists.sip-router.org
> Subject: sr-users Digest, Vol 90, Issue 72
> To: sr-users(a)lists.sip-router.org
> Date: Wed, 21 Nov 2012 08:58:40 +0100
>
> Send sr-users mailing list submissions to
> sr-users(a)lists.sip-router.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> or, via email, send a message with subject or body 'help' to
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>
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>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of sr-users digest..."
>
>
> Today's Topics:
>
> 1. Re: Shared Call Appearances module (Daniel-Constantin Mierla)
> 2. Re: Shared Call Appearances module (Andrew Mortensen)
> 3. Re: LCR weight (Juha Heinanen)
> 4. problem with $time Pseudo-Variables on 3.3.2 (Uri Shacked)
> 5. Re: Dialog module not decrementing profile size
> (Grant Bagdasarian)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Tue, 20 Nov 2012 22:56:10 +0100
> From: Daniel-Constantin Mierla <miconda(a)gmail.com>
> Subject: Re: [SR-Users] Shared Call Appearances module
> To: Ovidiu Sas <osas(a)voipembedded.com>
> Cc: "SIP Router - Kamailio \(OpenSER\) and SIP Express Router \(SER\)
> - Users Mailing List" <sr-users(a)lists.sip-router.org>
> Message-ID: <50ABFC7A.2070602(a)gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>
> On 11/20/12 10:41 PM, Ovidiu Sas wrote:
> > On Tue, Nov 20, 2012 at 4:22 PM, Andrew Mortensen
> > <admorten(a)isc.upenn.edu> wrote:
> >> On Nov 20, 2012, at 3:16 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote:
> >>
> >>> On 11/20/12 8:43 PM, Andrew Mortensen wrote:
> >>>> On Nov 20, 2012, at 4:08 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote:
> >>>>
> >>>>> Hello,
> >>>>>
> >>>>> thank for contribution!
> >>>>>
> >>>>> On 11/20/12 12:16 AM, Andrew Mortensen wrote:
> >>>>>> On Nov 19, 2012, at 5:53 PM, David | StyleFlare <david(a)styleflare.com> wrote:
> >>>>>>
> >>>>>>> Also quick read at the readme, it looks like it does not support multi-domain setups?
> >>>>>> Not yet, no. I don't think it would take much work, though.
> >>>>> I see it binds to usrloc, what information needs from there? We have two such modules right now, it would be good to make it work with both, kamailio's one has more features in regard to GRUU and partial outbound extensions.
> >>>> It's registering for contact expiration and deletion events. The callback terminates subscriptions for the expired/deleted contact. This should probably be configurable.
> >>> Ok, so it is about when a contact record in removed from location.
> >>>
> >>> Just to avoid misunderstandings, do you consider to make configurable this termination of subscriptions based on location?
> >> Yes. I can imagine a setup where the location data wouldn't be saved on the host handling SCA.
> >>
> >>> Also, you need only the AoR and contact address in a callback for location record removal event (un-registration or registration expiration), right?
> >> Correct, although Ovidiu makes a good point about (well-behaved) clients unsubscribing at the time of unregistration. In registering for usrloc callbacks, I'm trying to handle a couple problems. First, I'd like to avoid unnecessarily sending and retransmitting NOTIFYs to offline clients. More importantly, I want to be able to hear about expired registrations so I can clear appearance state on the rest of the subscribers.
> > For clearing appearance state, you should rely on the dialog module
> > and register a callback on the dialog module. When a dialog expires,
> > you can clear up the stale appearances.
> > Also, by enabling the sst module, you can have a tight control over
> > the duration of the dialog.
> > This will make the implementation cleaner and more robust.
> >
> I will not jump to such conclusions.
>
> While dialog module is designed to track active calls, does not mean
> everything has to be on top of it. Just for example, dispatcher module
> has an embedded lightweight call tracking system.
>
> No idea here what the broadsoft extensions require, afaik, there were
> some custom specs. It is no problem to have a dedicated dialog tracking
> system that is better designed for a particular functionality. We do
> have other duplicated systems, for example least cost routing, nat
> traversal, a.s.o.
>
> Also, optimizations and reuse of other parts can be done later if found
> to be better ways.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Tue, 20 Nov 2012 17:17:38 -0500
> From: Andrew Mortensen <admorten(a)isc.upenn.edu>
> Subject: Re: [SR-Users] Shared Call Appearances module
> To: Ovidiu Sas <osas(a)voipembedded.com>
> Cc: "SIP Router - Kamailio \(OpenSER\) and SIP Express Router \(SER\)
> - Users Mailing List" <sr-users(a)lists.sip-router.org>
> Message-ID: <65EAE745-8966-4848-870E-6A50246E0AB7(a)isc.upenn.edu>
> Content-Type: text/plain; charset=iso-8859-1
>
>
> On Nov 20, 2012, at 3:43 PM, Ovidiu Sas <osas(a)voipembedded.com> wrote:
>
> > Hello Andrew,
> >
> > First of all, thank you for sharing your work.
> > I was following this thread and I have a couple of questions. Why do
> > you need to bind to the usrloc module? The subscription itself should
> > be sufficient because if a phone will unregister, it will also
> > unsubscribe,
>
> Yes, that's good point. However, if the phone goes off-line instead of unregistering, the contact's registration will expire, and no unsubscribe will take place. As I mentioned in my reply to Daniel, I'm most concerned about catching this so I can release any appearances seized by the expired/deleted contact, and notify other members of the group. I'm certainly open to alternatives.
>
> > which leads me to the following question: why not add a
> > new dedicated module for call-info presence and reuse the existing
> > infrastructure from kamailio for handling subscriptions/presence. In
> > this case, your module should just push PUBLISH events to the presence
> > server and the server will automatically send out notifications for
> > subscribers that subscribed to call-info events.
>
> We've worked extensively with the existing presence & pua modules, albeit primarily dialog;sla using OpenSIPS behind the proxy. (Thanks, Anca!)
>
> SCA's unusual entanglement with call processing made me hesitate about building on top of the existing presence module. For a first pass, I also felt working with an entirely distinct module gave me more control and transparency during development of a very loosely documented event package, especially as I became more familiar with the available API.
>
> I don't have any objection to revisiting design decisions, of course. I'm sure the module will continue to evolve, and it would be nice to eliminate redundancy if possible.
>
> > Based on your README file, you inspect SIP requests/replies and based
> > on the presence of the call-info header, you create call-info events.
> > This is great for sharing the appearances between phones, but how do
> > you perform the retrieval of a call that was put on hold? Are you
> > using a dedicated B2BUA behind kamailio?
>
> No B2BUAs are involved.
>
> The INVITE retrieving a call held by another member of an SCA group has a particular set of characteristics: RURI, To and From URIs are all the SCA AoR; new unconfirmed dialog (no to-tag); and a Call-Info header referring to the index of the held call. The module detects this type of INVITE, looks up the dialog associated with the information in the Call-Info header, and injects a Replaces header with the dialog of the held call before relaying it to the remote party. The remote party must support RFC3891. I've only worked with Polycoms and Cisco gateways to this point, and both do support that RFC.
>
> I'm very interested in hearing from users with Cisco (and Snom and Aastra?) SCA setups. I have no doubt they'll find bugs resulting from assumptions I've made in the code because I've only tested with Polycom, not least the dependency on RFC3891 support.
>
> Thanks for your feedback!
>
> Best,
> andrew
>
>
>
> > On Tue, Nov 20, 2012 at 3:16 PM, Daniel-Constantin Mierla
> > <miconda(a)gmail.com> wrote:
> >>
> >> On 11/20/12 8:43 PM, Andrew Mortensen wrote:
> >>>
> >>> On Nov 20, 2012, at 4:08 AM, Daniel-Constantin Mierla <miconda(a)gmail.com>
> >>> wrote:
> >>>
> >>>> Hello,
> >>>>
> >>>> thank for contribution!
> >>>>
> >>>> On 11/20/12 12:16 AM, Andrew Mortensen wrote:
> >>>>>
> >>>>> On Nov 19, 2012, at 5:53 PM, David | StyleFlare <david(a)styleflare.com>
> >>>>> wrote:
> >>>>>
> >>>>>> Also quick read at the readme, it looks like it does not support
> >>>>>> multi-domain setups?
> >>>>>
> >>>>> Not yet, no. I don't think it would take much work, though.
> >>>>
> >>>> I see it binds to usrloc, what information needs from there? We have two
> >>>> such modules right now, it would be good to make it work with both,
> >>>> kamailio's one has more features in regard to GRUU and partial outbound
> >>>> extensions.
> >>>
> >>> It's registering for contact expiration and deletion events. The callback
> >>> terminates subscriptions for the expired/deleted contact. This should
> >>> probably be configurable.
> >>
> >>
> >> Ok, so it is about when a contact record in removed from location.
> >>
> >> Just to avoid misunderstandings, do you consider to make configurable this
> >> termination of subscriptions based on location?
> >>
> >> Also, you need only the AoR and contact address in a callback for location
> >> record removal event (un-registration or registration expiration), right?
> >>
> >>
> >>>
> >>>> Would you consider adding it to main git repository and maintaining it
> >>>> there?
> >>>
> >>> Yes, that would be great.
> >>
> >>
> >> We will prepare write access for you.
> >>
> >>
> >>> Would you prefer to have it in modules? I started work on it in
> >>> modules_s, but there's no particular reason it has to stay there.
> >>
> >> To make it work with both variants of usrloc modules, I think it requires
> >> some split of the code. IIRC, both modules still use same names for
> >> structures, so including files from both of them will fail.
> >>
> >> One solutions I am thinking of is to add two small modules, one that binds
> >> to usloc(k) and the other to usrloc(s). Each of these two modules will
> >> export a function that allow registering callbacks for execution when a
> >> location record expires, giving the aor, contact and event type. The
> >> existing module will use the intermediate module instead of directly binding
> >> to usrloc structures.
> >>
> >> Usrloc module from kamailio has more enhancements in regards to standard
> >> extensions for location services, still it lacks the feature of being able
> >> to store variables per contact record. It is the reason that kept back the
> >> removal (upon merging) of usrloc module from ser. The solution with
> >> intermediary modules should work until we have a merged usrloc module.
> >>
> >> What do you think about this proposal?
> >>
> >> Cheers,
> >> Daniel
> >>
> >>
> >> --
> >> Daniel-Constantin Mierla - http://www.asipto.com
> >> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> >>
>
>
>
>
> ------------------------------
>
> Message: 3
> Date: Wed, 21 Nov 2012 01:04:52 +0200
> From: Juha Heinanen <jh(a)tutpro.com>
> Subject: Re: [SR-Users] LCR weight
> To: "SIP Router - Kamailio \(OpenSER\) and SIP Express Router \(SER\)
> - Users Mailing List" <sr-users(a)lists.sip-router.org>
> Message-ID: <20652.3220.931000.738348(a)sip.test.fi>
> Content-Type: text/plain; charset=us-ascii
>
> Camila Troncoso writes:
>
> > scripts are provided in lcr/utils directory that can be used to check
> > the probabilities resulting from a given set of weight values. Same can
> > be done with command 'kamctl eval_weights'.
> >
> > I really don't understand what is the probability finally assign to each
> > gateway when they have the same priority.
> > All I can see is that it doesn't correspond to percentage of capacity from
> > the example above , but how can I determine what value gives the percent
> > that I want?
> >
> > Please give an example like:
> >
> > Rule_id Gw_id Priority weight
> > 36 68 1 60
> > 36 69 1 14
> > 36 70 1 13
> > 36 71 1 13
>
> you have to figure that out yourself using the script referenced in
> readme.
>
> -- juha
>
>
>
> ------------------------------
>
> Message: 4
> Date: Wed, 21 Nov 2012 09:55:17 +0200
> From: Uri Shacked <ushacked(a)gmail.com>
> Subject: [SR-Users] problem with $time Pseudo-Variables on 3.3.2
> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> Users Mailing List" <sr-users(a)lists.sip-router.org>
> Message-ID:
> <CAMMbDhQn1r_ke72-u2is8MyRXUV6KkLnZqYwxL5eZG44yYG_2Q(a)mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
>
> I upgraded to 3.3.2
>
> On 3.2 i used the variable $time() to get the current time for some process.
>
> Every 2 minutes a RTIMER process started, checked the $time(hour) and
> $time(min) and on a certain time executed something.
>
> This process stopped working on 3.3.2
>
> When i debug, i see that the $time() variable keeps on giving me the same
> hour and minute that was given on the first run.
>
> Meaning the $time() variable does not give the current time.
>
> I notice also that the $Ts that on 3.2 gave a current time stopped doing it
> and $TS does it now....
>
> Any help here?
>
> Is it OK and i need to check time another way? Or something in the $time()
> has changed...?
>
> Thanks,
>
> Uri
>
