date:20130828

Re: [SR-Users] unregister user when kamailio looses TCP connection.

2013-08-28 Thread Juha Heinanen

Vitaliy Aleksandrov writes:

> Didn't know about $T_reply_rid variable and that unregister can remove 
> only single contact.
> In my case the problem with unregister is that stale contact will be 
> removed only if somebody tries to call to a disconnected phone.

so you get one call to unregistered contact and after that it is not
anymore in location db.

is that not acceptable to you?  are you proposing to scan all contacts
for broken tcp connections on some timer or what?

-- juha

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Re: [SR-Users] unregister user when kamailio looses TCP connection.

2013-08-28 Thread Vitaliy Aleksandrov


On 08/28/2013 06:45 PM, Juha Heinanen wrote:

Vitaliy Aleksandrov writes:


If anybody else except me need this It would be great to fix known
problems and add it to kamailio.

i don't know if this come already up, but why not use this in branch
failure route:

unregister("location", "", "$T_reply_rid");
Didn't know about $T_reply_rid variable and that unregister can remove 
only single contact.
In my case the problem with unregister is that stale contact will be 
removed only if somebody tries to call to a disconnected phone.


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[SR-Users] NAT function

2013-08-28 Thread Kethzer Docteur

I need help to fixed my NAT configuration I have nat module load but nat
configuration is not working. ANY HELP PLease

#!ifdef WITH_NAT
loadmodule "nathelper.so"
loadmodule "rtpproxy.so"
loadmodule "nat_traversal.so"
#!endif


#!ifdef WITH_NAT
# - rtpproxy params -
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")

# - nathelper params -
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:pinger@67.215.8.130")

# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif


route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
add_contact_alias();
}
setflag(FLT_NATS);
}
#!endif
return;
}

# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;

rtpproxy_manage();

if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
add_contact_alias();
}
}
#!endif
return;
}

-- 
Kethzer Docteur
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Re: [SR-Users] Loopback

2013-08-28 Thread Marc Soda

Thanks, I appreciate it.

In this setup the there are 2 endpoints (700 and 701) peered up to an
Asterisk server (172.16.60.6) via a Kamailio proxy (172.16.60.20).  700
(172.16.60.28) is calling 701 (172.16.3.65).  When 701 answers the OK is
sent to the proxy and then to Asterisk.  Asterisk is then ACKing the OK.
 The ACK is being sent to the proxy and then the proxy should be sending it
back to the endpoint.  It is not.  The ACK is being sent to the proxy and
then the proxy is sending to itself again, via the loopback interface.  I
believe loose_route() should be re-writing the destination to be the
endpoint, but it not.

Trace:

U 172.16.60.28:54936 -> 172.16.60.20:5060
INVITE sip:7...@eng-reg1.example.com SIP/2.0.
Via: SIP/2.0/UDP 172.16.60.28:54936
;rport;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44.
Max-Forwards: 70.
From: ;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: .
Contact: .
Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi.
CSeq: 13862 INVITE.
Route: .
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS.
Supported: replaces, 100rel, timer, norefersub.
Session-Expires: 1800.
Min-SE: 90.
User-Agent: CSipSimple_mecha-15/r2272.
Proxy-Authorization: Digest username="sip700_tbs", realm="sip700_tbs",
nonce="Uh39eVId/E2Vxz5hgWC/7jMNGAf7rxrV", uri="sip:7...@eng-reg1.example.com",
response="8a74a8727baa45df84ea1374cb6668f2",
cnonce="EYGYpa1zWmEXcOMighUzGZ20cY2HJ7AJ", qop=auth, nc=0001.
Content-Type: application/sdp.
Content-Length:   340.
.
v=0.
o=- 3586685645 3586685645 IN IP4 172.16.60.28.
s=pjmedia.
c=IN IP4 172.16.60.28.
t=0 0.
m=audio 4004 RTP/AVP 99 0 8 101.
c=IN IP4 172.16.60.28.
a=rtcp:4005 IN IP4 172.16.60.28.
a=sendrecv.
a=rtpmap:99 SILK/24000.
a=fmtp:99 useinbandfec=0.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.


U 172.16.60.20:5060 -> 172.16.60.28:54936
SIP/2.0 100 trying -- your call is important to us.
Via: SIP/2.0/UDP 172.16.60.28:54936
;rport=54936;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44.
From: ;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: .
Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi.
CSeq: 13862 INVITE.
Server: kamailio (4.0.3 (x86_64/linux)).
Content-Length: 0.
.


U 172.16.60.20:5060 -> 172.16.60.6:5060
INVITE sip:7...@eng-reg1.example.com SIP/2.0.
Record-Route:
.
Via: SIP/2.0/UDP 172.16.60.20;branch=z9hG4bKe51f.d2338795.0.
Via: SIP/2.0/UDP 172.16.60.28:54936
;rport=54936;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44.
Max-Forwards: 16.
From: ;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: .
Contact: .
Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi.
CSeq: 13862 INVITE.
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS.
Supported: replaces, 100rel, timer, norefersub.
Session-Expires: 1800.
Min-SE: 90.
User-Agent: CSipSimple_mecha-15/r2272.
Content-Type: application/sdp.
Content-Length:   340.
.
v=0.
o=- 3586685645 3586685645 IN IP4 172.16.60.28.
s=pjmedia.
c=IN IP4 172.16.60.28.
t=0 0.
m=audio 4004 RTP/AVP 99 0 8 101.
c=IN IP4 172.16.60.28.
a=rtcp:4005 IN IP4 172.16.60.28.
a=sendrecv.
a=rtpmap:99 SILK/24000.
a=fmtp:99 useinbandfec=0.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.


U 172.16.60.6:5060 -> 172.16.60.20:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
172.16.60.20;branch=z9hG4bKe51f.d2338795.0;received=172.16.60.20;rport=5060.
Via: SIP/2.0/UDP 172.16.60.28:54936
;rport=54936;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44.
Record-Route:
.
From: ;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: .
Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi.
CSeq: 13862 INVITE.
Server: Asterisk1.8.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: .
Content-Length: 0.
.


U 172.16.60.6:5060 -> 172.16.60.20:5060
INVITE sip:sip701_tbs@172.16.60.20:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.16.60.6:5060;branch=z9hG4bK0919ead7;rport.
Max-Forwards: 70.
From: "Alpha" ;tag=as5e1a80d8.
To: .
Contact: .
Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060.
CSeq: 102 INVITE.
User-Agent: Asterisk1.8.
Date: Wed, 28 Aug 2013 13:34:03 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 263.
.
v=0.
o=root 1276916964 1276916964 IN IP4 172.16.60.6.
s=Asterisk PBX 1.8.15-cert2.
c=IN IP4 172.16.60.6.
t=0 0.
m=audio 21930 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.


U 172.16.60.20:5060 -> 172.16.60.6:5060
SIP/2.0 100 trying -- your call is important to us.
Via: SIP/2.0/UDP 172.16.60.6:5060;branch=z9hG4bK0919ead7;rport=5060.
From: "Alpha" ;tag=as5e1a80d8.
To: .
Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060.
CSeq: 102 INVITE.
Server: kamailio (4.0.3 (x86_64/linux)).
Content-Length: 0.
.


U 172.16.60.20:5060 -> 172.16.3.65:5060

[SR-Users] unregister user when kamailio looses TCP connection.

2013-08-28 Thread Juha Heinanen

Vitaliy Aleksandrov writes:

> If anybody else except me need this It would be great to fix known 
> problems and add it to kamailio.

i don't know if this come already up, but why not use this in branch
failure route:

unregister("location", "", "$T_reply_rid");

-- juha

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Re: [SR-Users] Loopback

2013-08-28 Thread Daniel-Constantin Mierla

If it is eth0 of the same server, then is the kernel sending via 
loopback interface at it detects the destination is itself.


You should paste here ngrep with the sip traffic from invite to bye in 
order to give more hints about what is going wrong there.


Cheers,
Daniel

On 8/28/13 3:09 PM, Marc Soda wrote:

I think I found my missing ACKs!  Can anyone tell me why they work be being 
sent to the loopback interface?  The destination address is still the external 
(eth0) IP.


--

Marc Soda, Sr. Systems Engineer
*CoreDial, LLC* | www.coredial.com 

1787 Sentry Parkway West, Building 16, Suite 100, Blue Bell, PA 19422
Office: (215) 297-4400 x203 | Fax: (215) 297-4401 | Email: 
ms...@coredial.com 


- - - - -

The information transmitted is intended only for the person or entity 
to which it is addressed and may contain confidential and/or 
privileged material. Any review, retransmission,  dissemination or 
other use of, or taking of any action in reliance upon, this 
information by persons or entities other than the intended recipient 
is prohibited. If you received this in error, please contact the 
sender and delete the material from any computer.



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--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Trainings - Berlin, Oct 21-24; Miami, Nov 11-13, 2013
  - more details about Kamailio trainings at http://www.asipto.com -

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[SR-Users] Loopback

2013-08-28 Thread Marc Soda

I think I found my missing ACKs!  Can anyone tell me why they work be
being sent to the loopback interface?  The destination address is
still the external (eth0) IP.



-- 

Marc Soda, Sr. Systems Engineer
*CoreDial, LLC* | www.coredial.com

1787 Sentry Parkway West, Building 16, Suite 100, Blue Bell, PA 19422
Office: (215) 297-4400 x203 | Fax: (215) 297-4401 | Email:
ms...@coredial.com

- - - - -

The information transmitted is intended only for the person or entity to
which it is addressed and may contain confidential and/or privileged
material. Any review, retransmission,  dissemination or other use of, or
taking of any action in reliance upon, this information by persons or
entities other than the intended recipient is prohibited. If you received
this in error, please contact the sender and delete the material from any
computer.
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[SR-Users] [Event] Kamailio at Astricon 2013 - discount code

2013-08-28 Thread Daniel-Constantin Mierla


Hello,

if anyone here is considering to attend Astricon, the discount code 
AC13DIGI gives 20% off.


  - http://www.astricon.net

The offer won't last too long and the seats are filling up quickly, so 
better hurry up. It's the biggest chance so far to learn about 
Kamailio+Asterisk together as well as meet the people behind the two 
projects in one place.


Cheers,
Daniel

On 8/26/13 6:36 PM, Daniel-Constantin Mierla wrote:

Hello,

I just want to announce that we will have a consistent presence this 
year at Astricon - it is the 10th edition, taking place in Atlanta, 
GA, USA, during October 8-10, 2013.


Using Kamailio and Asterisk together is very common and we want to 
make sure everyone uses properly the best of each. There will be 
several presentation from Kamailio developers as well as presence in 
expo area for demos and group chats. You can read more details at:
  - 
http://www.kamailio.org/w/2013/08/astricon-2013-the-10-years-celebration/


For details about the event itself, head directly to:
  - http://www.astricon.net

Looking forward to meeting many of you at Astricon!

Cheers,
Daniel



--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Trainings - Berlin, Oct 21-24; Miami, Nov 11-13, 2013
  - more details about Kamailio trainings at http://www.asipto.com -


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Re: [SR-Users] Using kamailio to play early media announcements

2013-08-28 Thread Carsten Bock

Hi,

a typical usecase for

rtpproxy_stream2uac(prompt_name, count),
rtpproxy_stop_stream2uac()

is "MusicOnHold". In this case, Kamailio does not have to create a new
SDP, but it would only insert the RTPProxy to play the prompt.

Kind regards,
Carsten

2013/8/28 Adnan <112linuxstockh...@gmail.com>:
> Thanks Olle. I know it's like asking MySQL to implement a picture filter ..
> :-)  but I am still wondering in what scenario do we use the following two
> functions from the rtpproxy module:
>
> rtpproxy_stream2uac(prompt_name, count),
> rtpproxy_stop_stream2uac()
>
> I am using the latest kamailio 4.0.3 and rtpproxy is running on the same
> machine and have encoded the voice prompt using makeann that comes with the
> rtpproxy.
>
> Would it playback the prompt as early media if kamailio on receiving the
> INVITE from the phone, replies with 183 Ringing with some manually crafted
> suitable sdp attached, can I then use some the above functions to playback
> the prompt as early media?
>
> I know it works with SEMs and others media gws and granted that it is a
> functionality not expected from a proxy but I am simply interested in
> answering the question whether it is possible with latest kamailio and
> rtpproxy and there is no other requirement restricting one when it comes to
> the configuration.
>
> If early media playback is not possible then what are the above rtpproxy
> functions for?
>
> /Adnan
>
> On Sat, Aug 17, 2013 at 12:27 PM, Olle E. Johansson  wrote:
>>
>>
>> 16 aug 2013 kl. 16:48 skrev Adnan <112linuxstockh...@gmail.com>:
>>
>> Hi,
>>
>> I know that this is not the purpose of the proxy but there is a situation
>> in which our kamailio needs to playback early media for a small number of
>> calls.
>>
>> Which module will be most suitable for this?
>>
>>
>> One way is to use rtpproxy and just functions such asrtpproxy_stream2...
>> to play back the prompts to the caller.
>>
>> Note that the caller is not behind NAT or behind any symmetric firewall.
>> So we a free to choose.
>>
>> We know we could do it easily by dispatching to asterisk where we use the
>> Progress() app or similar for freeswitch but we don't want add the overhead
>> of a new media server installation and configuration.
>>
>> We need a kamailio based one box solution.
>>
>> Then you have to write your own module, since Kamailio by design does not
>> handle media. Period.
>>
>> It's like asking MySQL to implement a picture filter for filters stored in
>> the database, it's just not in the design. Kamailio is a SIP server that
>> works with many media servers, but it's not a media server.
>>
>> Having said that you can propably fake a 183 and have a media server like
>> SEMS do it, but adding Asterisk is pretty much the same.
>>
>> /O
>>
>>
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>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
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>



-- 
Carsten Bock
CEO (Geschäftsführer)

ng-voice GmbH
Schomburgstr. 80
D-22767 Hamburg / Germany

http://www.ng-voice.com
mailto:cars...@ng-voice.com

Office +49 40 34927219
Fax +49 40 34927220

Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
Geschäftsführer: Carsten Bock
Ust-ID: DE279344284

Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
http://www.ng-voice.com/imprint/

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Re: [SR-Users] Using kamailio to play early media announcements

2013-08-28 Thread Adnan

Thanks Olle. I know it's like asking MySQL to implement a picture filter ..
:-)  but I am still wondering in what scenario do we use the following two
functions from the rtpproxy module:

rtpproxy_stream2uac(prompt_name, count),
rtpproxy_stop_stream2uac()

I am using the latest kamailio 4.0.3 and rtpproxy is running on the same
machine and have encoded the voice prompt using makeann that comes with the
rtpproxy.

Would it playback the prompt as early media if kamailio on receiving the
INVITE from the phone, replies with 183 Ringing with some manually crafted
suitable sdp attached, can I then use some the above functions to playback
the prompt as early media?

I know it works with SEMs and others media gws and granted that it is a
functionality not expected from a proxy but I am simply interested in
answering the question whether it is possible with latest kamailio and
rtpproxy and there is no other requirement restricting one when it comes to
the configuration.

If early media playback is not possible then what are the above rtpproxy
functions for?

/Adnan

On Sat, Aug 17, 2013 at 12:27 PM, Olle E. Johansson  wrote:

>
> 16 aug 2013 kl. 16:48 skrev Adnan <112linuxstockh...@gmail.com>:
>
> Hi,
>
> I know that this is not the purpose of the proxy but there is a situation
> in which our kamailio needs to playback early media for a small number of
> calls.
>
> Which module will be most suitable for this?
>
>
> One way is to use rtpproxy and just functions such asrtpproxy_stream2...
> to play back the prompts to the caller.
>
> Note that the caller is not behind NAT or behind any symmetric firewall.
> So we a free to choose.
>
> We know we could do it easily by dispatching to asterisk where we use the
> Progress() app or similar for freeswitch but we don't want add the overhead
> of a new media server installation and configuration.
>
> We need a kamailio based one box solution.
>
> Then you have to write your own module, since Kamailio by design does not
> handle media. Period.
>
> It's like asking MySQL to implement a picture filter for filters stored in
> the database, it's just not in the design. Kamailio is a SIP server that
> works with many media servers, but it's not a media server.
>
> Having said that you can propably fake a 183 and have a media server like
> SEMS do it, but adding Asterisk is pretty much the same.
>
> /O
>
>
> ___
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> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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Re: [SR-Users] Can I use kamailio replace of asterisk?

2013-08-28 Thread Gertjan Wolzak

Hello,

 

You can't replace Asterisk with Kamailio, well. depends on what you do with
Asterisk.

 

But remember Kamailio is a sip proxy server, Asterisk is a B2BUA.

 

As soon as Asterisk does  your voicemail, transcoding, etc. Kamailio is not
an option.

 

Just google Kamailio vs Asterisk, you will find more and better
explanations.

 

Good luck.

 

 

 

 

From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of ??,??!
Sent: woensdag 28 augustus 2013 2:59
To: sr-users
Subject: [SR-Users] Can I use kamailio replace of asterisk?

 

Hello, I have a question about the load balancer module of kamailio.

As the site http://kb.asipto.com/ say, Kamailio is as a SIP proxy router to
scale Asterisk.

 

Can I run a kamailio instance as the load balancer, and other several
instances as voice service replace of Asterisk?

 

If I can do that, could you give me a tutoral? We are using kamailio as our
server. Thank you very much.

--

Best Regards!

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Re: [SR-Users] Documentation for ims_registrar_pcscf

2013-08-28 Thread Carsten Bock

Hi,

I wrote a small Kamailio-IMS Installation howto in the Kamailio-Wiki:
=> http://www.kamailio.org/wiki/tutorials/ims/installation-howto

Maybe that will help you.

Kind regards,
Carsten

2013/8/27 Sai Krishna Kota :
> Hello,
>
> Can any one please provide the documentation for integrating PCSCF with
> Kamailio.
>
> The link for the documentation for ims_registrar_pcscf is not found. (
> http://kamailio.org/docs/modules/devel/modules/ims_registrar_pcscf.html)
>
> Please tell me what are the modules are to be loaded before pcscf.
>
>
> Thank you,
>
> Kota Sai Krishna
>
>
>
>
> ___
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> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



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Re: [SR-Users] hanging active dialog...

2013-08-28 Thread Gertjan Wolzak

Daniel,

 

Thank you for the examples.

 

Sorry, for the direct email, was not my intention.

 

Tried, and I get my results.

 

Thanks again.

 

Rgds,

 

Gertjan

 

From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] 
Sent: vrijdag 23 augustus 2013 17:31
To: Gertjan Wolzak; 'Kamailio (SER) - Users Mailing List'
Subject: Re: [SR-Users] hanging active dialog...

 

The example is the raw command to send, but kamctl takes care to format it.
You should do:

kamctl fifo profile_list_dlg callers

or


kamctl fifo profile_list_dlg callers _URI_

Replace the _URI_ with the value that was in $fu.

Cheers,
Daniel

On 8/23/13 3:34 PM, Gertjan Wolzak wrote:

I have tried with kamctl...

 

But because I found the documentation not very clear, I think I am doing it
wrong.

 

Lets take the 8.6:

 

:profile_list_dlgs:_reply_fifo_file_

   inbound_calls

   _empty_line_

 

Where the parameters are profile and value, whereby the value is optional.

 

I tried : kamctl fifo :profile_list_dlgs: _reply_fifo_file_inbound_calls
callers

 

Im pretty sure that is not correct.

 

Been looking for the correct why on google, but can find a clear
explanation.

 

Gertjan.

 

From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
Daniel-Constantin Mierla
Sent: vrijdag 23 augustus 2013 15:06
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] hanging active dialog...

 

Have you listed the profile via kamctl or kamcmd? The commands are in the
readme:

http://kamailio.org/docs/modules/stable/modules/dialog.html#idp3737432

Is the afferent profile listing any dialog?

Cheers,
Daniel

On 8/23/13 2:50 PM, Gertjan Wolzak wrote:

Hello Carlos,

 

I have been looking at your module, cant wait to work with it, especially
the prepaid part.

 

But as this system is in production, cant just start experimenting, want
to. But cant.

 

So just have to wait for some tips on the cli.

 

But thanks.

 

Rgds,

 

Gertjan 

 

From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Carlos Ruiz Díaz
Sent: vrijdag 23 augustus 2013 14:37
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] hanging active dialog...

 

I don't know what exactly happened in your case but if you don't have time
to investigate you can try cnxcc module [1] channel control to achieve the
same goal.

 

Take a look at the sample configuration file located in [2]

 

 
xlog("L_INFO", "Setting up channel based 
 credit control");
 
 
 $var(max_chan)   = 2;
 
 $var(retcode)= cnxcc_set_max_channels("$var(client)",
"$var(max_chan)");
 
 
 if ($var(retcode) == -1) {
 
 xlog("Error setting up credit control");
 return;
 
 }
 
 
$var(count) = -1;
 
 
if (!cnxcc_get_channel_count("$var(client)", "$var(count)")) {
 
xlog("Error getting customer's channel 
count"
);
 
}
 
 
xlog("L_INFO", "CNXCC ROUTE: $var(client) has $var(count) call(s)");
 
 
 if ($var(retcode) < -1) {
 
 xlog("Too many channels for <
 /span>custome
 r");
 
 sl_send_reply(403, "Forbidden");
 
 
 if (!cnxcc_terminate_all("$var(client)")) {
 
  xlog("Error terminating customer's calls");
 
 }
 
 
 exit;
 }
 
 
[1] http://kamailio.org/docs/modules/devel/modules/cnxcc.html
[2]
https://github.com/caruizdiaz/cnxcc/blob/master/example/kamailio-cnxcc.cfg
 
Regards,
Carlos
 

 

On Fri, Aug 23, 2013 at 5:16 AM, Gertjan Wolzak  wrote:

 

Goodmorning All,

 

I use the following route to check for concurrent calls by the same user, if
a concurrent call is tried it is not allowed.

 

route[CONCURRENT]

{

xlog("SCRIPT: Conccurrent call check");

 

if(!get_profile_size("caller","$fu","$avp(nrcalls)"))

{

sl_send_reply("403", "Call not matching profile");

exit;

}

xlog("SCRIPT: caller value for $fu is $avp(nrcalls)");

if($avp(nrcalls)>= 1)

{

sl_send_reply("403", "Active calls limit exceeded");

exit;

}

dlg_manage();

if(!set_dlg_profile("caller","$fu"))

{

sl_send_reply("500", "No new channels in this profile");

exit;

}

 

xlog("SCRIPT: caller value for $fu is now $avp(nrcalls)");

 

}

 

Now I had a situation where a user could not call because the
get_profile_size for this user gave the value 1. So another call was not
allowed.

But the user did not have a call active. As no dialogs were active, checked
that.

 

I assume there can be a lot of reasons why this happens, also I want to use
the I dont want to know the cause method to solve this.

 

So I looked at the dialog module documentation

[SR-Users] Can I use kamailio replace of asterisk?

2013-08-28 Thread ????????????

Hello, I have a question about the load balancer module of kamailio.
 As the site http://kb.asipto.com/ say, Kamailio is as a SIP proxy router to 
scale Asterisk.
  
 Can I run a kamailio instance as the load balancer, and other several 
instances as voice service replace of Asterisk?
  
 If I can do that, could you give me a tutoral? We are using kamailio as our 
server. Thank you very much.
 
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[SR-Users] ims_registrar_pcscf

2013-08-28 Thread Sai Krishna Kota

 

Hello,

Can any one please provide the documentation for integrating PCSCF with
Kamailio.

The link for the documentation for ims_registrar_pcscf is not found. (
http://kamailio.org/docs/modules/devel/modules/ims_registrar_pcscf.html)

Please tell me what are the modules are to be loaded before pcscf.


Thank you,

Kota Sai Krishna

 

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[SR-Users] Documentation for ims_registrar_pcscf

2013-08-28 Thread Sai Krishna Kota

Hello,

Can any one please provide the documentation for integrating PCSCF with
Kamailio.

The link for the documentation for ims_registrar_pcscf is not found. (
http://kamailio.org/docs/modules/devel/modules/ims_registrar_pcscf.html)

Please tell me what are the modules are to be loaded before pcscf.


Thank you,

Kota Sai Krishna

 

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Re: [SR-Users] Documentation for ims_registrar_pcscf

Re: [SR-Users] hanging active dialog...

[SR-Users] Can I use kamailio replace of asterisk?

[SR-Users] ims_registrar_pcscf

[SR-Users] Documentation for ims_registrar_pcscf

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