date:20111027

Re: [SR-Users] How to limit maximum number of calls

2011-10-27 Thread Austin Einter

Hi
I did change kamailio.cfg as below and getting some error.

*//Loaded dialog module*
*loadmodule "dialog.so"*

My route block looks as below.

*# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
route {*
*# per request initial checks
route(REQINIT);
setflag(FLT_NATS);
# NAT detection
route(NAT);*

*# handle requests within SIP dialogs
route(WITHINDLG);*

*if (is_method("INVITE"))
{
if($DLG_count > 1)
{
sl_send_reply("503","RESOURCE UNAVAILABLE");
exit;
}
}
*
I added the if block shown in red color.

When I start the kamailio proxy I get below error.

[root@www1 kamailio]#
[root@www1 kamailio]# kamailio -T -E -n 1 -l 174.37.8.178 -l 127.0.0.1 -W
epoll_et -l udp:174.37.8.178:26588
loading modules under
/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/
Listening on
 udp: 174.37.8.178:26588
 udp: 127.0.0.1:26588
Aliases:
 udp: localhost:26588
 udp: localhost.localdomain:26588
 udp: 174.37.8.178-static.reverse.softlayer.com:26588
 0(26959) INFO: usrloc [hslot.c:53]: locks array size 512
 0(26959) ERROR: dialog [dialog.c:435]: no dlg flag set!!
 0(26959) ERROR:  [sr_module.c:875]: init_mod(): Error while
initializing module dialog (/usr/local/lib/kamailio/modules_k/dialog.so)
ERROR: error while initializing modules
[root@www1 kamailio]#
[root@www1 kamailio]#

It looks , dialog module facing problem during initialization.

I am bit new to this, not getting any clue . Kindly let me know whats going
wrong.

Thanks
Austin

On Fri, Oct 28, 2011 at 9:53 AM, Alex Balashov wrote:

> On 10/27/2011 11:26 PM, Austin Einter wrote:
>
> Do you mean that I need to modify kamailio.cfg to have max calls
>> limitation?
>>
>
> Quitely so.
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
>  __**_
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> sr-users@lists.sip-router.org
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Re: [SR-Users] How to limit maximum number of calls

2011-10-27 Thread Alex Balashov


On 10/27/2011 11:26 PM, Austin Einter wrote:


Do you mean that I need to modify kamailio.cfg to have max calls
limitation?


Quitely so.

--
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Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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Re: [SR-Users] How to limit maximum number of calls

2011-10-27 Thread Austin Einter

Thanks Alex
Do you mean that I need to modify kamailio.cfg to have max calls limitation?

Regards
Kamal

On Fri, Oct 28, 2011 at 8:47 AM, Alex Balashov wrote:

> Investigate the "dialog" module.
>
> --
> This message was painstakingly thumbed out on my mobile, so apologies for
> brevity, errors, and general sloppiness.
>
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
> On Oct 27, 2011, at 9:01 PM, Austin Einter 
> wrote:
>
> > Hi
> > I am running kamailio proxy (1.5) as an intermediate proxy, where all my
> SIP signalling packets are passing through.
> > I want to limit maximum number of calls kamailio proxy server can handle
> at a time.
> > How can I do this, please give me some pointer in this regard.
> >
> > Regards
> > Austin
> > ___
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Re: [SR-Users] How to limit maximum number of calls

2011-10-27 Thread Alex Balashov

Investigate the "dialog" module.

--
This message was painstakingly thumbed out on my mobile, so apologies for 
brevity, errors, and general sloppiness.

Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

On Oct 27, 2011, at 9:01 PM, Austin Einter  wrote:

> Hi
> I am running kamailio proxy (1.5) as an intermediate proxy, where all my SIP 
> signalling packets are passing through.
> I want to limit maximum number of calls kamailio proxy server can handle at a 
> time.
> How can I do this, please give me some pointer in this regard.
>  
> Regards
> Austin
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Re: [SR-Users] How to limit maximum number of calls

2011-10-27 Thread Austin Einter

One correction. I am using Kamailio 3.1.5.



On Fri, Oct 28, 2011 at 8:31 AM, Austin Einter wrote:

> Hi
> I am running kamailio proxy (1.5) as an intermediate proxy, where all my
> SIP signalling packets are passing through.
> I want to limit maximum number of calls kamailio proxy server can handle
> at a time.
> How can I do this, please give me some pointer in this regard.
>
> Regards
> Austin
>
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[SR-Users] How to limit maximum number of calls

2011-10-27 Thread Austin Einter

Hi
I am running kamailio proxy (1.5) as an intermediate proxy, where all my SIP
signalling packets are passing through.
I want to limit maximum number of calls kamailio proxy server can handle
at a time.
How can I do this, please give me some pointer in this regard.

Regards
Austin
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Re: [SR-Users] Kamailio3.2 debian repo fails

2011-10-27 Thread Daniel-Constantin Mierla


Hello,

indeed, it looks like debs for 3.2 are not yet generated -- the apt repo 
for it has not debs. Jon (cc-ed) was taking care of it, afaik, he is 
traveling these days, so it may take a bit until he can loot at it.


Meanwhile, if you have a debian system where you can install the 
dependencies, you can build he debs yourself. In the sources directory 
of kamailio do:


ln -s pkg/kamailio/deb/squeeze debian
make deb

Debs will be dumped in parent directory.

Cheers,
Daniel

On 10/27/11 2:37 PM, Benjamin Henrion wrote:

Hi,

I have the following /etc/apt/sources.list:


$ cat /etc/apt/sources.list
deb http://ftp.debian.org/debian squeeze main contrib non-free
deb-src http://ftp.debian.org/debian squeeze main contrib non-free

deb http://deb.kamailio.org/kamailio32 squeeze main
deb-src http://deb.kamailio.org/kamailio32 squeeze main


Which generates the following error:

$ apt-get update
Ign http://deb.kamailio.org squeeze Release.gpg
Ign http://deb.kamailio.org/kamailio32/ squeeze/main Translation-en
Ign http://deb.kamailio.org/kamailio32/ squeeze/main Translation-en_US
Ign http://deb.kamailio.org/kamailio32/ squeeze/main Translation-en_US.UTF-8
Ign http://deb.kamailio.org squeeze Release
Ign http://deb.kamailio.org squeeze/main Sources
Hit http://ftp.debian.org squeeze Release.gpg
Ign http://ftp.debian.org/debian/ squeeze/contrib Translation-en
Ign http://ftp.debian.org/debian/ squeeze/contrib Translation-en_US
Ign http://deb.kamailio.org squeeze/main amd64 Packages
Ign http://deb.kamailio.org squeeze/main Sources
Ign http://ftp.debian.org/debian/ squeeze/contrib Translation-en_US.UTF-8
Ign http://ftp.debian.org/debian/ squeeze/main Translation-en
Ign http://ftp.debian.org/debian/ squeeze/main Translation-en_US
Ign http://ftp.debian.org/debian/ squeeze/main Translation-en_US.UTF-8
Ign http://ftp.debian.org/debian/ squeeze/non-free Translation-en
Ign http://ftp.debian.org/debian/ squeeze/non-free Translation-en_US
Ign http://ftp.debian.org/debian/ squeeze/non-free Translation-en_US.UTF-8
Hit http://ftp.debian.org squeeze Release
Ign http://deb.kamailio.org squeeze/main amd64 Packages
Err http://deb.kamailio.org squeeze/main Sources
   404  Not Found
Err http://deb.kamailio.org squeeze/main amd64 Packages
   404  Not Found
Hit http://ftp.debian.org squeeze/main Sources
Hit http://ftp.debian.org squeeze/contrib Sources
Hit http://ftp.debian.org squeeze/non-free Sources
Hit http://ftp.debian.org squeeze/main amd64 Packages
Hit http://ftp.debian.org squeeze/contrib amd64 Packages
Hit http://ftp.debian.org squeeze/non-free amd64 Packages
W: Failed to fetch
http://deb.kamailio.org/kamailio32/dists/squeeze/main/source/Sources.gz
  404  Not Found
W: Failed to fetch
http://deb.kamailio.org/kamailio32/dists/squeeze/main/binary-amd64/Packages.gz
  404  Not Found
E: Some index files failed to download, they have been ignored, or old
ones used instead.


Who is responsible for the debian packaging?

Best,

--
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FFII Brussels - +32-484-566109 - +32-2-4148403
"In July 2005, after several failed attempts to legalise software
patents in Europe, the patent establishment changed its strategy.
Instead of explicitly seeking to sanction the patentability of
software, they are now seeking to create a central European patent
court, which would establish and enforce patentability rules in their
favor, without any possibility of correction by competing courts or
democratically elected legislators."

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Re: [SR-Users] Asterisk 1.8 and Kamailio 1.5 issue

2011-10-27 Thread Daniel-Constantin Mierla


Hello,

you have to provide the sip trace taken on the sip server, in order to 
see what is received and what is sent out by kamailio. Looks like the 
one you pasted here is from client.


You can use ngrep on kamailio server:

ngrep -d any -qt -W byline port 5060

Also, the packets you pasted next are from two different calls (see the 
call-id header). The second seems to be completed ok, but something is 
not good for asterisk and it sends bye. Maybe you can spot something in 
the logs of asterisk.


Cheers,
Daniel

On 10/26/11 8:25 PM, Rowell Rufino wrote:

Hi,
We are having issues where the "OK" or "ACK" is that is coming from 
the phone is not relayed by OpenSER to Asterisk.
Below is the sip trace...  I am also attaching a tcpdump. Please help 
what we can do.


Received from udp:10.1.10.80:5060  at 
26/10/2011 10:22:41:476 (490 bytes):


SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 
10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060
From: "Virgil Menendez" >;tag=6wkdms1r20
To: ;user=phone>;tag=as0b87218f

Call-ID: 3c26755bf15c-9iq08xqqblo6
CSeq: 4 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH

Supported: replaces, timer
Content-Length: 0



Sent to udp:10.1.10.80:5060  at 26/10/2011 
10:22:41:481 (387 bytes):


ACK sip:vm9513261429@10.1.10.83:5060 
 SIP/2.0

v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport
Route: 
f: "Virgil Menendez" >;tag=6wkdms1r20
t: ;user=phone>;tag=as0b87218f

i: 3c26755bf15c-9iq08xqqblo6
CSeq: 4 ACK
Max-Forwards: 70
m: http://sip:91421@10.30.0.64:5060>>;reg-id=1
l: 0



Received from udp:10.1.10.80:5060  at 
26/10/2011 10:22:42:130 (868 bytes):


SIP/2.0 200 OK
Via: SIP/2.0/UDP 
10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060

Record-Route: 
From: "Virgil Menendez" >;tag=qi3i8ze6z8
To: ;user=phone>;tag=as3f8c0f96

Call-ID: 3c2676547a8d-2t5yi6jok1sv
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH

Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: >

Content-Type: application/sdp
Content-Length: 256

v=0
o=root 1355451627 1355451627 IN IP4 10.1.10.83
s=Asterisk PBX 1.8.7.1
c=IN IP4 10.1.10.83
t=0 0
m=audio 16094 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv



Sent to udp:10.1.10.80:5060  at 26/10/2011 
10:22:42:132 (385 bytes):


ACK sip:9513261429@10.1.10.83:5060 
 SIP/2.0

v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport
Route: 
f: "Virgil Menendez" >;tag=qi3i8ze6z8
t: ;user=phone>;tag=as3f8c0f96

i: 3c2676547a8d-2t5yi6jok1sv
CSeq: 2 ACK
Max-Forwards: 70
m: http://sip:91421@10.30.0.64:5060>>;reg-id=1
l: 0



Received from udp:10.1.10.80:5060  at 
26/10/2011 10:22:42:232 (503 bytes):


BYE sip:91421@10.30.0.64:5060  SIP/2.0
Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0
Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1
Max-Forwards: 69
From: ;user=phone>;tag=as3f8c0f96
To: "Virgil Menendez" >;tag=qi3i8ze6z8

Call-ID: 3c2676547a8d-2t5yi6jok1sv
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.7.1
*X-Asterisk-HangupCause: Protocol error, unspecified
*X-Asterisk-HangupCauseCode: 111
Content-Length: 0

Regards,

Rowell


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Re: [SR-Users] T1 timer

2011-10-27 Thread Daniel-Constantin Mierla




On 10/27/11 6:15 PM, Vitaliy Aleksandrov wrote:
Thank you for your quick reply. t_set_retr doesn't affects transaction 
created by uac_req_send.


The second way proposed by you works great (tm settings for uac, 
t_set_retr for t_relay).


Just to be clear, you have tested t_set_retr only in route block, not in 
event_route[tm:local-request], right?


Cheers,
Daniel


Hello,

On 10/26/11 12:10 PM, Vitaliy wrote:

Hello,

How can i change T1 timeout for transaction created by the UAC 
module (uac_req_send).
Do i have to call t_set_retr before uac_req_send and then 
t_reset_retr() before t_relay ?


I use uac_req_send to send accounting to the remote server and i 
want to increase retransmit timeout for such signalling messages.


it is not a way dedicated for uac_req_send() and I haven't played 
with variants -- you can try what you wrote above and if does not 
work, try to set the timer parameters for tm to the one you want for 
uac and use for forwarded transactions t_set_retr. One other way to 
try is to make a event_route[tm:local-request] and use there the 
t_set_retr.


Let me know if any of it works, we can add to docs if so, otherwise I 
will look for implementing a solution for it.


Cheers,
Daniel




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Re: [SR-Users] Dispatcher Confusion (v3.2.0)

2011-10-27 Thread Daniel-Constantin Mierla


Hello,

On 10/27/11 5:30 PM, Asgaroth wrote:

Hi Daniel,

[...]

Since with 3.2 seemed that it was lost capability to go inactive after
a certain number of failures (ds_probing_threshold), there is a new
state 'trying' that can be used for it. Means that you can set a
destination in trying state couple of times and then it becomes
inactive. In 3.1 it was using a confusing mechanism based on probing mode.

Can you explain this trying state and "lost capability to go inactive
after certain number of failures" a little more please and how it
relates to the new trying->inactive states. I would like to understand
how these states relate so that I can test better.


I was not using the feature in the past, but from the source code I 
could see that there was a way not to go directly in probing mode (which 
in the past meant not to select the gateway anymore), but just count 
failure until a threshold is reached and then set probing.


So if threshold was 3 and there were (in 3.1.x-):
ds_mark_dst(p) => state still active (no probing, gateway still selected)
ds_mark_dst(p) => state still active (no probing, gateway still selected)
ds_mark_dst(p) => state goes to probing (inactive, gateway not selected)

Now (3.3.x+), since probing can be always on, even for active 
destinations (to detect when they go down), you can get previous like 
behavior with trying state:


ds_mark_dst(t) => state trying (gateway still selected)
ds_mark_dst(t) => state trying (gateway still selected)
ds_mark_dst(t) => state goes to inactive (gateway not selected)

Default failure counter threshold is 1, so goes to inactive as soon as 
you set trying, but you can change it via ds_probing_threshold parameter.







So right now there are states: active, inactive, trying and disabled,
plus modes: probing, not-probing. A destination can be selected only
if it is active or trying. It will not be selected in inactive and
disabled. Probing mode specifies whether keepalives should be sent to
destinations, can be done per address or globally with the module
parameter ds_probing_mode. If a keepalive is not replied, the address
is marked as trying first and later will become inactive if keeps
being non-responsive.

OK, so if I understand this above paragraph correctly, if I have
ds_probing_mode = 0, then I need to set mode manually to probing for a
gateway that has failed "ds_probing_threshold" times? If a server times
out and I set state/mode to "ip", then I assume probing will commence.
In this case the server will not responde to probe requests (as it has
crashed), does this mean then that the state will change to "trying"
because there was no probe response recieved from destination?
Probing is no longer a gw selection state, but a mode switch to send 
keepalives or not to a gateway. So if you want these keepalives and 
ds_probing_mode=0, you have to set 'p' in any of the states you want 
keepalives. A matter of the reply code from keepalives, the state in 
probing mode is changed to active if it is 200ok or a reply code 
configured in module parameter, or to trying if it is a failure (which 
may end up in inactive when failure threshold is met). ds_probing_mode 
controls as well if a keepalive reply will maintain the probing mode or not.


Cheers,
Daniel

--
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Re: [SR-Users] T1 timer

2011-10-27 Thread Vitaliy Aleksandrov

Thank you for your quick reply. t_set_retr doesn't affects transaction 
created by uac_req_send.


The second way proposed by you works great (tm settings for uac, 
t_set_retr for t_relay).

Hello,

On 10/26/11 12:10 PM, Vitaliy wrote:

Hello,

How can i change T1 timeout for transaction created by the UAC module 
(uac_req_send).
Do i have to call t_set_retr before uac_req_send and then 
t_reset_retr() before t_relay ?


I use uac_req_send to send accounting to the remote server and i want 
to increase retransmit timeout for such signalling messages.


it is not a way dedicated for uac_req_send() and I haven't played with 
variants -- you can try what you wrote above and if does not work, try 
to set the timer parameters for tm to the one you want for uac and use 
for forwarded transactions t_set_retr. One other way to try is to make 
a event_route[tm:local-request] and use there the t_set_retr.


Let me know if any of it works, we can add to docs if so, otherwise I 
will look for implementing a solution for it.


Cheers,
Daniel




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Re: [SR-Users] Dispatcher Confusion (v3.2.0)

2011-10-27 Thread Asgaroth

Hi Daniel,

On 27/10/2011 15:57, Daniel-Constantin Mierla wrote:
> Hello,
>
> I just pushed to remote GIT repository in master branch a bit of
> refactoring about the states and ds_mark_dst().

Thanks, I will test the dev branch in a short while and get back to you.

>
> Since with 3.2 seemed that it was lost capability to go inactive after
> a certain number of failures (ds_probing_threshold), there is a new
> state 'trying' that can be used for it. Means that you can set a
> destination in trying state couple of times and then it becomes
> inactive. In 3.1 it was using a confusing mechanism based on probing mode.

Can you explain this trying state and "lost capability to go inactive
after certain number of failures" a little more please and how it
relates to the new trying->inactive states. I would like to understand
how these states relate so that I can test better.

>
>
> So right now there are states: active, inactive, trying and disabled,
> plus modes: probing, not-probing. A destination can be selected only
> if it is active or trying. It will not be selected in inactive and
> disabled. Probing mode specifies whether keepalives should be sent to
> destinations, can be done per address or globally with the module
> parameter ds_probing_mode. If a keepalive is not replied, the address
> is marked as trying first and later will become inactive if keeps
> being non-responsive.

OK, so if I understand this above paragraph correctly, if I have
ds_probing_mode = 0, then I need to set mode manually to probing for a
gateway that has failed "ds_probing_threshold" times? If a server times
out and I set state/mode to "ip", then I assume probing will commence.
In this case the server will not responde to probe requests (as it has
crashed), does this mean then that the state will change to "trying"
because there was no probe response recieved from destination?

>
> The parameter for ds_mark_dst() can be now a combination between
> states and probing mode, like ds_mark_dst("ip").

Excellent, thank you, that will help :)

>
> Maybe you can give it a try and let me know if it works -- I was not
> able to test in my side yet -- remember this id master branch (devel
> version 3.3.0-devX). Once this is ok and the internal states are clear
> and acting properly, I will look into backporting the fixes to 3.2
> branch.

OK, I will try this in a short while and get back to you.

Thanks

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Re: [SR-Users] sdpops and dynamic codec ids

2011-10-27 Thread Daniel-Constantin Mierla


Hello,

I committed on master branch the code for support of dynamic codecs ids:

http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=511dc62e6a6ca74324f42b66a23bd9d80b377252

I tested and seemed ok, but could try many options, so if you can give 
it a try as well would be good before backporting. I may be offline due 
to traveling, so if all ok and you need it in 3.2 quickly, feel free to 
cherry pick it to stable branch.


Thanks,
Daniel


On 10/22/11 8:25 PM, Daniel-Constantin Mierla wrote:

Hello,

Right now is working only for non-dynamic codecs ids, indeed. There 
was a bug in the sdp parser that prevented implementing it in the 
first place, but when I fixed that before the 3.2.0 release, I forgot 
about this one. I will try to add it asap.


Cheers,
Daniel

On 10/22/11 3:20 PM, Juha Heinanen wrote:

i made call sdp_keep_codecs_by_name("PCMU,PCMA,speex"); on an invite
request and it didn't keep speex although it was in the sdp:

 Session Description Protocol
 Session Description Protocol Version (v): 0
 Owner/Creator, Session Id (o): sems 1 1 IN IP4 
192.98.102.10

 Session Name (s): sems
 Connection Information (c): IN IP4 192.98.102.10
 Time Description, active time (t): 0 0
 Media Description, name and address (m): audio 4 
RTP/AVP 101 102 8 0 3 2 9 96 97 98 99 100

 Media Attribute (a): rtpmap:101 iLBC/8000
 Media Attribute (a): rtpmap:102 speex/8000
 Media Attribute (a): rtpmap:8 PCMA/8000
 Media Attribute (a): rtpmap:0 PCMU/8000
 Media Attribute (a): rtpmap:3 GSM/8000
 Media Attribute (a): rtpmap:2 G721/8000
 Media Attribute (a): rtpmap:9 g722/8000
 Media Attribute (a): rtpmap:96 telephone-event/8000
 Media Attribute (a): rtpmap:97 G726-32/8000
 Media Attribute (a): rtpmap:98 G726-24/8000
 Media Attribute (a): rtpmap:99 G726-40/8000
 Media Attribute (a): rtpmap:100 G726-16/8000

it is so that sdpops module does not support codecs, whose ids are
dynamic, but only codecs with static ids?

-- juha

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Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat
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Re: [SR-Users] Dispatcher Confusion (v3.2.0)

2011-10-27 Thread Daniel-Constantin Mierla


Hello,

I just pushed to remote GIT repository in master branch a bit of 
refactoring about the states and ds_mark_dst().


Since with 3.2 seemed that it was lost capability to go inactive after a 
certain number of failures (ds_probing_threshold), there is a new state 
'trying' that can be used for it. Means that you can set a destination 
in trying state couple of times and then it becomes inactive. In 3.1 it 
was using a confusing mechanism based on probing mode.


So right now there are states: active, inactive, trying and disabled, 
plus modes: probing, not-probing. A destination can be selected only if 
it is active or trying. It will not be selected in inactive and 
disabled. Probing mode specifies whether keepalives should be sent to 
destinations, can be done per address or globally with the module 
parameter ds_probing_mode. If a keepalive is not replied, the address is 
marked as trying first and later will become inactive if keeps being 
non-responsive.


The parameter for ds_mark_dst() can be now a combination between states 
and probing mode, like ds_mark_dst("ip").


Maybe you can give it a try and let me know if it works -- I was not 
able to test in my side yet -- remember this id master branch (devel 
version 3.3.0-devX). Once this is ok and the internal states are clear 
and acting properly, I will look into backporting the fixes to 3.2 branch.


Cheers,
Daniel

On 10/26/11 11:15 PM, Asgaroth wrote:

Hi Daniel,

On 26/10/2011 18:17, Daniel-Constantin Mierla wrote:

if you tried with 3.2.x, it was the case, since I just backported from
master branch the commit I did to sort out better the behaviour based
on probing state. Try again now with latest 3.2 branch.


Thanks, the changes you made there for 3.2.x bring the behaviour into
the same operation as the devel branch. My findings so far are listed below:

Kamailio Version:

# sbin/kamailio -V
version: kamailio 3.2.0 (i386/linux) 7c241c
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: 7c241c
compiled on 18:36:47 Oct 26 2011 with gcc 4.1.2

All the testing observations below is done with ds_probing_mode = 0 (A
gateway will only be "pinged" when it is set into probing mode). I
assume that the probing state needs to be manually set using fifo
command or ds_mark_dst()/ds_mark_dst("s") command.

Now I understand what you have been saying about the differentiation
between active/inactive/disabled state and probing enabled between the
states. However, I am still unable to set a destination into
inactive-probing state from within the routing script.

I understand that a gateway will only be selected when the state of the
destination gateway is in AX (Active) or AP (Active-Probing) state.
These states can be achieved by calling ds_mark_dst("a") for AX (Active)
state, or ds_mark_dst("p") for AP (Active-Probing) state from the
routing script.

I understand that a gateway will *not* be selected when the state of the
distination is in DX (Disabled), IX (Inactive) or IP (Inactive-Probing)
state. Only one of the three states in this case can be achieved via
routing script, IX (Inactive) can be achieved by calling ds_mark_dst()
or ds_mark_dst("i"). DX (Disabled) state can be achieved by calling
ds_set_state fifo command, this makes sense as it would be an
administrative down. However, IP (Inactive-Probing) cannot be called
from routing script, which, I think, is essential in some scenarios, for
example, say a destination crashes for some reason, and you want to
probe the gateway for when it becomes available again but you dont want
to use it while it is down, then you need to be able to set the state
from routing script to IP (Inactive-Probing).

To sum it all up, the states from fifo command all work as advertised,
the following states (AX/AP/IX) can be achieved from routing script, IP
state cannot be achieved from routing script (but can be achieved from
fifo command).

Disabled state is set manually using rpc/fifo command.

Is it intentional to not be able to set state to IP from routing script,
or is this also something that needs to be looked at?

Thanks for all the help thus far :)

Thanks

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--
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Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat
http://linkedin.com/in/miconda -- http://twitter.com/miconda


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SIP Exp

[SR-Users] Kamailio3.2 debian repo fails

2011-10-27 Thread Benjamin Henrion

Hi,

I have the following /etc/apt/sources.list:


$ cat /etc/apt/sources.list
deb http://ftp.debian.org/debian squeeze main contrib non-free
deb-src http://ftp.debian.org/debian squeeze main contrib non-free

deb http://deb.kamailio.org/kamailio32 squeeze main
deb-src http://deb.kamailio.org/kamailio32 squeeze main


Which generates the following error:

$ apt-get update
Ign http://deb.kamailio.org squeeze Release.gpg
Ign http://deb.kamailio.org/kamailio32/ squeeze/main Translation-en
Ign http://deb.kamailio.org/kamailio32/ squeeze/main Translation-en_US
Ign http://deb.kamailio.org/kamailio32/ squeeze/main Translation-en_US.UTF-8
Ign http://deb.kamailio.org squeeze Release
Ign http://deb.kamailio.org squeeze/main Sources
Hit http://ftp.debian.org squeeze Release.gpg
Ign http://ftp.debian.org/debian/ squeeze/contrib Translation-en
Ign http://ftp.debian.org/debian/ squeeze/contrib Translation-en_US
Ign http://deb.kamailio.org squeeze/main amd64 Packages
Ign http://deb.kamailio.org squeeze/main Sources
Ign http://ftp.debian.org/debian/ squeeze/contrib Translation-en_US.UTF-8
Ign http://ftp.debian.org/debian/ squeeze/main Translation-en
Ign http://ftp.debian.org/debian/ squeeze/main Translation-en_US
Ign http://ftp.debian.org/debian/ squeeze/main Translation-en_US.UTF-8
Ign http://ftp.debian.org/debian/ squeeze/non-free Translation-en
Ign http://ftp.debian.org/debian/ squeeze/non-free Translation-en_US
Ign http://ftp.debian.org/debian/ squeeze/non-free Translation-en_US.UTF-8
Hit http://ftp.debian.org squeeze Release
Ign http://deb.kamailio.org squeeze/main amd64 Packages
Err http://deb.kamailio.org squeeze/main Sources
  404  Not Found
Err http://deb.kamailio.org squeeze/main amd64 Packages
  404  Not Found
Hit http://ftp.debian.org squeeze/main Sources
Hit http://ftp.debian.org squeeze/contrib Sources
Hit http://ftp.debian.org squeeze/non-free Sources
Hit http://ftp.debian.org squeeze/main amd64 Packages
Hit http://ftp.debian.org squeeze/contrib amd64 Packages
Hit http://ftp.debian.org squeeze/non-free amd64 Packages
W: Failed to fetch
http://deb.kamailio.org/kamailio32/dists/squeeze/main/source/Sources.gz
 404  Not Found
W: Failed to fetch
http://deb.kamailio.org/kamailio32/dists/squeeze/main/binary-amd64/Packages.gz
 404  Not Found
E: Some index files failed to download, they have been ignored, or old
ones used instead.


Who is responsible for the debian packaging?

Best,

--
Benjamin Henrion 
FFII Brussels - +32-484-566109 - +32-2-4148403
"In July 2005, after several failed attempts to legalise software
patents in Europe, the patent establishment changed its strategy.
Instead of explicitly seeking to sanction the patentability of
software, they are now seeking to create a central European patent
court, which would establish and enforce patentability rules in their
favor, without any possibility of correction by competing courts or
democratically elected legislators."

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