date:20160211

Re: [OpenSIPS-Users] Topology Hiding Usage Question

2016-02-11 Thread Bogdan-Andrei Iancu


Hi Nathaniel,

When you do topology hiding, from SIP routing perspective, OpenSIPS will 
act as an end-point and not as a proxy. So, if you do TH, makes no sense 
to do record_route and loose_route and they are specific to proxy routing.

So,
1) no
2) what you mean "hide" ? to be replied from opensips ? you can do that 
if you are sure you do not break any end-2-end communication
3) where you put it in the script is up to you - just be sure that 
topology_hiding() function must be called only for initial INVITE requests.


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 11.02.2016 09:13, Nathaniel L. Keeling III wrote:

Hello,

I am implementing the topology_hiding module and had a couple of 
questions. After reading the tutorial on topology hiding. it states 
that you use the topology_hiding_match() function to check for 
topology hiding. In the tutorial, this function is performed after 
performing the has_totag() check and that one exist. Normally, at 
least from a generated script, the loose_route() check is performed.


1. Do we still need to perform the loose_route() check if we are 
performing the topology_hiding_match() check?
2. If you are using the dialog module, can it hide requests that are 
not in a dialogs?
3. Would it be ok to execute topology_hiding() function within the 
route(RELAY) or should it be done else where? I am using a generated 
script as my base.


Thanks

Nathaniel Keeling

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Re: [OpenSIPS-Users] Radius Async

2016-02-11 Thread Dragomir Haralambiev

Hello,

Problem with radius still exists.
Any idea?

Regards,
Dragomir

2016-02-05 18:58 GMT+02:00 Dragomir Haralambiev :

> Now my Makefile begin with follow lines:
>
> ifeq ("$(FREERADIUS)", "")
> ifneq ("$(wildcard /usr/include/freeradius-client.h)","")
> FREERADIUS=1
> endif
> endif
>
> I compile with :
> FREERADIUS=1 make all
> FREERADIUS=1 make install
>
> Problem still exists!
>
> 2016-02-05 16:28 GMT+02:00 Ionut Ionita :
>
>> Can you try this patch?
>> And when you compile try
>> FREERADIUS=1 make all
>>
>> Ionut Ionita
>> OpenSIPS Developer
>>
>> On 02/05/2016 04:08 PM, Dragomir Haralambiev wrote:
>>
>> I do this but problem is not resolved
>>
>>
>> 2016-02-05 15:27 GMT+02:00 Ionut Ionita :
>>
>>> Uncomment the first line
>>>
>>> FREERADIUS=1
>>> # freeradius libs check (must be done in toplevel makefile)
>>> ifneq ("$(wildcard /usr/include/freeradius-client.h)","")
>>> FREERADIUS=1
>>> else
>>> #FREERADIUS=0
>>> endif
>>>
>>> and compile OpenSIPS again. We will try to fix that because
>>> it's quite hard to configure.
>>>
>>> Ionut Ionita
>>> OpenSIPS Developer
>>>
>>> On 02/05/2016 03:23 PM, Dragomir Haralambiev wrote:
>>>
>>> Hi,
>>>
>>> Here is firs lines from my MakeFile
>>>
>>> #FREERADIUS=1
>>> # freeradius libs check (must be done in toplevel makefile)
>>> ifneq ("$(wildcard /usr/include/freeradius-client.h)","")
>>> FREERADIUS=1
>>> else
>>> #FREERADIUS=0
>>> endif
>>>
>>>
>>> I not see the /usr/include/freeradius-client.h
>>>
>>> Regards,
>>>
>>> 2016-02-05 15:01 GMT+02:00 Ionut Ionita < 
>>> ionution...@opensips.org>:
>>>
 Thank you for your time.
 Are you sure you have in your OpenSIPS Makefile FREERADIUS=1
 for using freeradius-client.h and not radiusclient-ng? It's the first
 commented line in the Makefile

 #FREERADIUS=1

  Because as I see in the output of nm all the async functions from
 OpenSIPS
  are there and also all the freeradius library async functions are
 there, so the
 used library is correct.

 Ionut Ionita
 OpenSIPS Developer

 On 02/05/2016 02:32 PM, Dragomir Haralambiev wrote:

 Here is tests

 2016-02-05 12:35 GMT+02:00 Ionut Ionita < 
 ionution...@opensips.org>:

> Do you have any other errors? Can you provide the log please?
> After this please do the following:
>
> go to module folder:
>
> cd modules/aaa_radius
>
> from here run the following:
>
> make proper #it's not an error; is just a warning that you should
> build it from the main folder
> FREERADIUS=1 NICER=0 make #here you will also receive a warning;
> please provide output
> nm aaa_radius.so # please provide the output of this
> nm `ldd aaa_radius.so | grep radius | awk '{print $3}'` #and also for
> this
>
> Thank you
>
> Ionut Ionita
> OpenSIPS Developer
>
> On 02/05/2016 11:51 AM, Dragomir Haralambiev wrote:
>
> make proper working ONLY in main dir
>
> I make follow:
> make clean
> make proper
> make all
> make install
>
> When try to run I receive:
>
>  unknown async command , missing loadmodule?
>
>
> Regards,
> Dragomir
>
> 2016-02-05 11:32 GMT+02:00 Ionut Ionita < 
> ionution...@opensips.org>:
>
>> The library looks ok. Could you try recompiling the module?
>>
>> cd modules/aaa_radius ; make proper
>>
>> and then recompile it from opensips main folder.
>>
>> Ionut Ionita
>> OpenSIPS Developer
>>
>> On 02/05/2016 11:22 AM, Dragomir Haralambiev wrote:
>>
>> Hi, Thnaks for your replay.
>>
>> I send patched freeradius-client.h.
>>
>> Regards,
>> Dragomir
>>
>> 2016-02-05 11:15 GMT+02:00 Ionut Ionita < 
>> ionution...@opensips.org>:
>>
>>> Ok! After the patch you should have in the header the following
>>> three lines
>>>
>>> #ifndef RADIUS_ASYNC_SUPPORT
>>> #define RADIUS_ASYNC_SUPPORT
>>> #endif
>>>
>>> at the very beginning. This helps compiling the async support
>>> for the aaa_radius module only if this patch is applied.
>>>
>>>
>>> Ionut Ionita
>>> OpenSIPS Developer
>>>
>>> On 02/05/2016 10:56 AM, Dragomir Haralambiev wrote:
>>>
>>> Do you want to send freeradius-client.h after the patch?
>>>
>>> 2016-02-05 10:49 GMT+02:00 Dragomir Haralambiev <
>>> goup2...@gmail.com>:
>>>
 Yes I apply the patch and then compile Opensips 2.2.

 2016-02-05 10:05 GMT+02:00 Ionut Ionita <
 ionution...@opensips.org>:

> Hi Dragomir,
>
> Did you apply the patch on the freeradius-client library as

Re: [OpenSIPS-Users] Call-id issue in Cancel message generated by tm / $T_fr_inv_timeout

2016-02-11 Thread Bogdan-Andrei Iancu


Hi Julian,

I will have to test this and come back to you.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 10.02.2016 17:45, Julian Santer wrote:

Hi guys,

we seem to got the same issue like John Nash on 2015/08/12.
We use OpenSips 2.1.2 with the latest revision from git repo.

Like John we are not sure if it is a bug or our mistake ;-)

We are using topology hiding and the Call ID in the CANCEL, generated 
by the TM module, is not the same, as the call ID in the initial INVITE.


The call flow looks like:
PSTN carrier -> gw-carrier (topo hiding) -> core (topo hiding) -> 
gw-consumer (topo-hiding) -> UAC consumer


The CANCEL generated by the TM module of the core, sended to the 
gw-consumer is rejected by the gw-consumer.


The CANCEL starts on the core. So let me show you
1) the initial INVITE, which the core receives from the gw-carrier 
(Call-ID: GW-CARRIER)
2) the initial INVITE, which the core and sends to the gw-consumer 
(Call-ID: Core)
3) the CANCEL generated by the core after $T_fr_inv_timeout (Call-ID: 
GW-CARRIER)


1)
INVITE sip:12345@IP_CORE SIP/2.0
Via: SIP/2.0/UDP IP_GW-CARRIER:5060;branch=z9hG4bK6aa2.7710f555.0
From: ;tag=E3AE5C5C-1A42
To: 
Call-ID: 
GW-CARRIER_EjJFKHdkNlktdGM2RV93ZV5MWHdlS0wvAn1HN14LYjFHLgRiXU1aHGdCWlcE

CSeq: 101 INVITE
Max-Forwards:  8
Remote-Party-ID: 
;party=calling;screen=yes;privacy=off

Contact: 
Expires: 180
Content-Type: application/sdp
Content-Length: 474
sdp ...

2)
INVITE sip:12345@IP_UAC:PORT_UAC SIP/2.0
Via: SIP/2.0/UDP IP_CORE:5060;branch=z9hG4bK45da.82f6fd55.0
Route: 
From: ;tag=E3AE5C5C-1A42
To: 
Call-ID: 
Core_ExwGCwAcKhgvdAgnFg58LwQGAXUOXSAzC3A1JFB/FCcWCWFzGAQkXHInPFQGDzYYI3oPCCAMahZeEy11JywlCVEG

CSeq: 101 INVITE
Max-Forwards:  7
Remote-Party-ID: 
;party=calling;screen=yes;privacy=off

Contact: 
Expires: 180
Content-Type: application/sdp
Content-Length: 426
sdp ...

3)
CANCEL sip:12345@IP_UAC:PORT_UAC SIP/2.0
Via: SIP/2.0/UDP IP_CORE:5060;branch=z9hG4bK45da.82f6fd55.0
From: ;tag=E3AE5C5C-1A42
Call-ID: 
GW-CARRIER_EjJFKHdkNlktdGM2RV93ZV5MWHdlS0wvAn1HN14LYjFHLgRiXU1aHGdCWlcE

To: 
CSeq: 101 CANCEL
Max-Forwards: 70
Route: 
Reason: SIP;cause=480;text="NO_ANSWER"
User-Agent: OpenSIPS (2.1.2 (x86_64/linux))
Content-Length: 0

Kind regards,
Julian Santer
Raiffeisen OnLine


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Re: [OpenSIPS-Users] [BOOK] Building Telephony Systems with OpenSIPS - 2.1 version

2016-02-11 Thread Bogdan-Andrei Iancu


Hi Schneur,

OK, I will check with the editor if they can push fixes to the book. 
Thanks for reporting it!


Thanks and Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 09.02.2016 13:20, Schneur Rosenberg wrote:


Chapter 1, page 11 (page 36 of the eBook) on the bottom of the page 
there is a initial invite of userA calling userB, and the contact is 
of userB, it should be userA


On Feb 8, 2016 3:13 PM, "Bogdan-Andrei Iancu" > wrote:


Hi Schneur,

Could you point the name of the chapter and the context of that
mistake, so I can double check it ?

Thanks & Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 07.02.2016 14:53, Schneur Rosenberg wrote:


I bought the book and I think I found the first mistake, in the
initial  invite packet on page 11, the contact header is of the
destination and not of the origination UAC

On Feb 4, 2016 3:37 PM, "Bogdan-Andrei Iancu"
> wrote:

Hello all,

Flavio Goncalves and I are happy to announce the publishing
of the second edition of "Building Telephony Systems with
OpenSIPS", covering OpenSIPS version 2.1 .

Also many thanks to the Packt Publishing house for making it
happened and to all our reviewers who help us to make this
book better.


https://www.packtpub.com/networking-and-servers/building-telephony-systems-opensips-second-edition

Enjoy !

-- 
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
http://www.opensips-solutions.com


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Re: [OpenSIPS-Users] PJSIP example configuration not working

2016-02-11 Thread Bogdan-Andrei Iancu


Hi Mateusz,

May I ask what is the connection of your question with OpenSIPS ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 09.02.2016 15:30, Mateusz Kowalski wrote:

Hello,

My problem is having Asterisk with PJSIP realtime (configured as 
https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip) not binding 
to port 5060 (or any other) by default and thus not listening for the clients.

My demo configuration looks as follows:

pjsip.conf
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

sorcery.conf

[res_pjsip]
endpoint=realtime,ps_endpoints
auth=realtime,ps_auths
aor=realtime,ps_aors
domain_alias=realtime,ps_domain_aliases
contact=realtime,ps_contacts

[res_pjsip_endpoint_identifier_ip]
identify=realtime,ps_endpoint_id_ips

extconfig.conf

[settings]
ps_endpoints => odbc,asterisk
ps_auths => odbc,asterisk
ps_aors => odbc,asterisk
ps_domain_aliases => odbc,asterisk
ps_endpoint_id_ips => odbc,asterisk
ps_contacts => odbc,asterisk

Unfortunately Google does not help at all. It is quite simple config, so I 
assumed it should work almost out-of-the-box. I've checked DB and it looks 
Asterisk can correctly connect and query data, the problem is with not binding 
the port itself.


Just if it my have impact, I have disabled module chan_sip completely.

Thanks for any help,
-- Mateusz


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Re: [OpenSIPS-Users] How to TLS ?

2016-02-11 Thread Hamid Hashmi

Nabeel
I dont know how to present a certificate from client. I have tried using Xoiper 
(Android - Free), SFLphone (Ubuntu) and CsipSimple (Android) but there was no 
options set a public key. 
Now I am using CA signed certificates in opensips with disabled flags of 
verify_cert and require_cert, having an error of 477 Send failed (477/TM). 
Hamid R. HashmiSoftware Engineer - VoIPVopium A/S

Date: Tue, 9 Feb 2016 08:48:41 +
From: nabeelshik...@gmail.com
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] How to TLS ?

Hi, 
Does the client present a client certificate? If not, then with 
modparam("proto_tls","require_cert", "1"), OpenSIPS misleadingly logs:

'failed to accept: rejected by client'.  What it actually means is that the 
client failed to present a certificate. 
On 9 Feb 2016 6:06 am, "Hamid Hashmi"  wrote:

It will be a great help if you please help me in configuring TLS. I have 
followed this to configure TLS but could not able to verify certificates.
its working if disable following flags
modparam("proto_tls","verify_cert", "0")modparam("proto_tls","require_cert", 
"0")
BUT not verifying certificates. Please see logs if enabled
modparam("proto_tls","verify_cert", "1")modparam("proto_tls","require_cert", 
"1")
then have following ERROR
Feb  9 05:57:14 comoyo-dev-ec2-siplb SIPLB[29867]: 
[udp:keepalive@192.168.26.181:8000]: Receive request OPTIONS from local server 
[192.168.26.181] 
Feb  9 05:57:14 comoyo-dev-ec2-siplb SIPLB[29868]: ERROR:proto_tls:tls_accept: 
New TLS connection from 115.186.93.1:47015 failed to accept: rejected by client
Feb  9 05:57:14 comoyo-dev-ec2-siplb SIPLB[29868]: 
ERROR:proto_tls:tls_read_req: failed to do pre-tls reading
Feb  9 05:57:17 comoyo-dev-ec2-siplb SIPLB[29863]: 
[tcp:siplb@192.168.26.180:6080]: In LOCAL Route sending OPTIONS to 
192.168.26.181 
Feb  9 05:57:17 comoyo-dev-ec2-siplb SIPLB[29863]: 
INFO:core:probe_max_sock_buff: using snd buffer of 244 kb
Feb  9 05:57:17 comoyo-dev-ec2-siplb SIPLB[29863]: 
INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 17Regards
Hamid R. Hashmi

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Re: [OpenSIPS-Users] Call-id issue in Cancel message generated by tm / $T_fr_inv_timeout

2016-02-11 Thread Julian Santer


Hi Bogdan,

thank you for your time. If you need further informations (config files etc.) 
let me know.

Kind regards,
Julian Santer
Raiffeisen OnLine

Am 11.02.2016 um 10:26 schrieb Bogdan-Andrei Iancu:

Hi Julian,

I will have to test this and come back to you.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 10.02.2016 17:45, Julian Santer wrote:

Hi guys,

we seem to got the same issue like John Nash on 2015/08/12.
We use OpenSips 2.1.2 with the latest revision from git repo.

Like John we are not sure if it is a bug or our mistake ;-)

We are using topology hiding and the Call ID in the CANCEL, generated by the TM 
module, is not the same, as the call ID in the initial INVITE.

The call flow looks like:
PSTN carrier -> gw-carrier (topo hiding) -> core (topo hiding) -> gw-consumer 
(topo-hiding) -> UAC consumer

The CANCEL generated by the TM module of the core, sended to the gw-consumer is 
rejected by the gw-consumer.

The CANCEL starts on the core. So let me show you
1) the initial INVITE, which the core receives from the gw-carrier (Call-ID: 
GW-CARRIER)
2) the initial INVITE, which the core and sends to the gw-consumer (Call-ID: 
Core)
3) the CANCEL generated by the core after $T_fr_inv_timeout (Call-ID: 
GW-CARRIER)

1)
INVITE sip:12345@IP_CORE SIP/2.0
Via: SIP/2.0/UDP IP_GW-CARRIER:5060;branch=z9hG4bK6aa2.7710f555.0
From: ;tag=E3AE5C5C-1A42
To: 
Call-ID: GW-CARRIER_EjJFKHdkNlktdGM2RV93ZV5MWHdlS0wvAn1HN14LYjFHLgRiXU1aHGdCWlcE
CSeq: 101 INVITE
Max-Forwards:  8
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact: 
Expires: 180
Content-Type: application/sdp
Content-Length: 474
sdp ...

2)
INVITE sip:12345@IP_UAC:PORT_UAC SIP/2.0
Via: SIP/2.0/UDP IP_CORE:5060;branch=z9hG4bK45da.82f6fd55.0
Route: 
From: ;tag=E3AE5C5C-1A42
To: 
Call-ID: 
Core_ExwGCwAcKhgvdAgnFg58LwQGAXUOXSAzC3A1JFB/FCcWCWFzGAQkXHInPFQGDzYYI3oPCCAMahZeEy11JywlCVEG
CSeq: 101 INVITE
Max-Forwards:  7
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact: 
Expires: 180
Content-Type: application/sdp
Content-Length: 426
sdp ...

3)
CANCEL sip:12345@IP_UAC:PORT_UAC SIP/2.0
Via: SIP/2.0/UDP IP_CORE:5060;branch=z9hG4bK45da.82f6fd55.0
From: ;tag=E3AE5C5C-1A42
Call-ID: GW-CARRIER_EjJFKHdkNlktdGM2RV93ZV5MWHdlS0wvAn1HN14LYjFHLgRiXU1aHGdCWlcE
To: 
CSeq: 101 CANCEL
Max-Forwards: 70
Route: 
Reason: SIP;cause=480;text="NO_ANSWER"
User-Agent: OpenSIPS (2.1.2 (x86_64/linux))
Content-Length: 0

Kind regards,
Julian Santer
Raiffeisen OnLine


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Re: [OpenSIPS-Users] Call-id issue in Cancel message generated by tm / $T_fr_inv_timeout

2016-02-11 Thread Bogdan-Andrei Iancu


Julian,

Please update from GIT repo and give it a new try. It should work now 
(at least it does for me).


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 11.02.2016 12:07, Julian Santer wrote:

Hi Bogdan,

thank you for your time. If you need further informations (config 
files etc.) let me know.


Kind regards,
Julian Santer
Raiffeisen OnLine

Am 11.02.2016 um 10:26 schrieb Bogdan-Andrei Iancu:

Hi Julian,

I will have to test this and come back to you.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 10.02.2016 17:45, Julian Santer wrote:

Hi guys,

we seem to got the same issue like John Nash on 2015/08/12.
We use OpenSips 2.1.2 with the latest revision from git repo.

Like John we are not sure if it is a bug or our mistake ;-)

We are using topology hiding and the Call ID in the CANCEL, 
generated by the TM module, is not the same, as the call ID in the 
initial INVITE.


The call flow looks like:
PSTN carrier -> gw-carrier (topo hiding) -> core (topo hiding) -> 
gw-consumer (topo-hiding) -> UAC consumer


The CANCEL generated by the TM module of the core, sended to the 
gw-consumer is rejected by the gw-consumer.


The CANCEL starts on the core. So let me show you
1) the initial INVITE, which the core receives from the gw-carrier 
(Call-ID: GW-CARRIER)
2) the initial INVITE, which the core and sends to the gw-consumer 
(Call-ID: Core)
3) the CANCEL generated by the core after $T_fr_inv_timeout 
(Call-ID: GW-CARRIER)


1)
INVITE sip:12345@IP_CORE SIP/2.0
Via: SIP/2.0/UDP IP_GW-CARRIER:5060;branch=z9hG4bK6aa2.7710f555.0
From: ;tag=E3AE5C5C-1A42
To: 
Call-ID: 
GW-CARRIER_EjJFKHdkNlktdGM2RV93ZV5MWHdlS0wvAn1HN14LYjFHLgRiXU1aHGdCWlcE

CSeq: 101 INVITE
Max-Forwards:  8
Remote-Party-ID: 
;party=calling;screen=yes;privacy=off

Contact: 
Expires: 180
Content-Type: application/sdp
Content-Length: 474
sdp ...

2)
INVITE sip:12345@IP_UAC:PORT_UAC SIP/2.0
Via: SIP/2.0/UDP IP_CORE:5060;branch=z9hG4bK45da.82f6fd55.0
Route: 
From: ;tag=E3AE5C5C-1A42
To: 
Call-ID: 
Core_ExwGCwAcKhgvdAgnFg58LwQGAXUOXSAzC3A1JFB/FCcWCWFzGAQkXHInPFQGDzYYI3oPCCAMahZeEy11JywlCVEG

CSeq: 101 INVITE
Max-Forwards:  7
Remote-Party-ID: 
;party=calling;screen=yes;privacy=off

Contact: 
Expires: 180
Content-Type: application/sdp
Content-Length: 426
sdp ...

3)
CANCEL sip:12345@IP_UAC:PORT_UAC SIP/2.0
Via: SIP/2.0/UDP IP_CORE:5060;branch=z9hG4bK45da.82f6fd55.0
From: ;tag=E3AE5C5C-1A42
Call-ID: 
GW-CARRIER_EjJFKHdkNlktdGM2RV93ZV5MWHdlS0wvAn1HN14LYjFHLgRiXU1aHGdCWlcE

To: 
CSeq: 101 CANCEL
Max-Forwards: 70
Route: 
Reason: SIP;cause=480;text="NO_ANSWER"
User-Agent: OpenSIPS (2.1.2 (x86_64/linux))
Content-Length: 0

Kind regards,
Julian Santer
Raiffeisen OnLine


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Re: [OpenSIPS-Users] Call-id issue in Cancel message generated by tm / $T_fr_inv_timeout

2016-02-11 Thread Julian Santer


Bogdan,

we tried now the latest GIT release and it works like a charm ;-)
Thank you for quick fix.

Kind regards,
Julian Santer
Raiffeisen OnLine

Am 11.02.2016 um 14:02 schrieb Bogdan-Andrei Iancu:

Julian,

Please update from GIT repo and give it a new try. It should work now (at least 
it does for me).

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 11.02.2016 12:07, Julian Santer wrote:

Hi Bogdan,

thank you for your time. If you need further informations (config files etc.) 
let me know.

Kind regards,
Julian Santer
Raiffeisen OnLine

Am 11.02.2016 um 10:26 schrieb Bogdan-Andrei Iancu:

Hi Julian,

I will have to test this and come back to you.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 10.02.2016 17:45, Julian Santer wrote:

Hi guys,

we seem to got the same issue like John Nash on 2015/08/12.
We use OpenSips 2.1.2 with the latest revision from git repo.

Like John we are not sure if it is a bug or our mistake ;-)

We are using topology hiding and the Call ID in the CANCEL, generated by the TM 
module, is not the same, as the call ID in the initial INVITE.

The call flow looks like:
PSTN carrier -> gw-carrier (topo hiding) -> core (topo hiding) -> gw-consumer 
(topo-hiding) -> UAC consumer

The CANCEL generated by the TM module of the core, sended to the gw-consumer is 
rejected by the gw-consumer.

The CANCEL starts on the core. So let me show you
1) the initial INVITE, which the core receives from the gw-carrier (Call-ID: 
GW-CARRIER)
2) the initial INVITE, which the core and sends to the gw-consumer (Call-ID: 
Core)
3) the CANCEL generated by the core after $T_fr_inv_timeout (Call-ID: 
GW-CARRIER)

1)
INVITE sip:12345@IP_CORE SIP/2.0
Via: SIP/2.0/UDP IP_GW-CARRIER:5060;branch=z9hG4bK6aa2.7710f555.0
From: ;tag=E3AE5C5C-1A42
To: 
Call-ID: GW-CARRIER_EjJFKHdkNlktdGM2RV93ZV5MWHdlS0wvAn1HN14LYjFHLgRiXU1aHGdCWlcE
CSeq: 101 INVITE
Max-Forwards:  8
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact: 
Expires: 180
Content-Type: application/sdp
Content-Length: 474
sdp ...

2)
INVITE sip:12345@IP_UAC:PORT_UAC SIP/2.0
Via: SIP/2.0/UDP IP_CORE:5060;branch=z9hG4bK45da.82f6fd55.0
Route: 
From: ;tag=E3AE5C5C-1A42
To: 
Call-ID: 
Core_ExwGCwAcKhgvdAgnFg58LwQGAXUOXSAzC3A1JFB/FCcWCWFzGAQkXHInPFQGDzYYI3oPCCAMahZeEy11JywlCVEG
CSeq: 101 INVITE
Max-Forwards:  7
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact: 
Expires: 180
Content-Type: application/sdp
Content-Length: 426
sdp ...

3)
CANCEL sip:12345@IP_UAC:PORT_UAC SIP/2.0
Via: SIP/2.0/UDP IP_CORE:5060;branch=z9hG4bK45da.82f6fd55.0
From: ;tag=E3AE5C5C-1A42
Call-ID: GW-CARRIER_EjJFKHdkNlktdGM2RV93ZV5MWHdlS0wvAn1HN14LYjFHLgRiXU1aHGdCWlcE
To: 
CSeq: 101 CANCEL
Max-Forwards: 70
Route: 
Reason: SIP;cause=480;text="NO_ANSWER"
User-Agent: OpenSIPS (2.1.2 (x86_64/linux))
Content-Length: 0

Kind regards,
Julian Santer
Raiffeisen OnLine


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Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

2016-02-11 Thread Podrigal, Aron

Hint: Do you listen on more then one ip or proto?
On Feb 11, 2016 7:44 PM, "surya"  wrote:

> Hi,
>
> My understanding about the *children* parameter in the cfg file is that the
> same number of child process will be created. For ex: if I write
> *children=1* only 1 child will be created and all the requests will be
> handled by same child. Please correct me if I am wrong.
>
> But, when I start the opensips presence server with *children=1* I see 6
> child inits, still all the requests are processed by one child (which had
> rank 1). Can someone please explain why other child are being init.
>
> Thanks,
> Surya
>
>
>
> --
> View this message in context:
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> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
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Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

2016-02-11 Thread Alex Balashov


Hi,

The command 'opensipsctl ps' should clear this up.

e.g. in my case, with children=8,

[root@sip ~]# opensipsctl ps
Process::  ID=0 PID=743932 Type=attendant
Process::  ID=1 PID=743933 Type=MI FIFO
Process::  ID=2 PID=743934 Type=time_keeper
Process::  ID=3 PID=743935 Type=timer
Process::  ID=4 PID=743936 Type=SIP receiver udp:x.x.x.x:5060
Process::  ID=5 PID=743937 Type=SIP receiver udp:x.x.x.x:5060
Process::  ID=6 PID=743938 Type=SIP receiver udp:x.x.x.x:5060
Process::  ID=7 PID=743939 Type=SIP receiver udp:x.x.x.x:5060
Process::  ID=8 PID=743940 Type=SIP receiver udp:x.x.x.x:5060
Process::  ID=9 PID=743941 Type=SIP receiver udp:x.x.x.x:5060
Process::  ID=10 PID=743942 Type=SIP receiver udp:x.x.x.x:5060
Process::  ID=11 PID=743943 Type=SIP receiver udp:x.x.x.x:5060

-- Alex

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[OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

2016-02-11 Thread surya

Hi,

My understanding about the *children* parameter in the cfg file is that the
same number of child process will be created. For ex: if I write
*children=1* only 1 child will be created and all the requests will be
handled by same child. Please correct me if I am wrong.

But, when I start the opensips presence server with *children=1* I see 6
child inits, still all the requests are processed by one child (which had
rank 1). Can someone please explain why other child are being init.

Thanks,
Surya



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Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

2016-02-11 Thread surya

Hi Aron,

Thanks for the reply.

I am just using UDP but with your hint, I rechecked the CFG and disabled the
tcp, set *disable_tcp=yes* and there are 4 child.
I am listening on single IP(I don't know how to do on multiple). :)


Thanks,
Surya



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Re: [OpenSIPS-Users] Lots of presence:p_tm_callback, 408 warning

2016-02-11 Thread surya

Hi Bogdan,

Thanks.

There must be a queue for the incoming requests though. Can you redirect me
to the code which queues the requests? I tried to look through the codes but
it's difficult.

Basically, I want to check how much time request spend in the queue and
calculate the response time for requests. I am facing some issues with the
response time measurements. 


Thanks,
Surya



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Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

2016-02-11 Thread surya

I am not opensips expert, therefore, my understanding may not be correct but
when I see the log, I see the 4 child init and the print statements from the
child_init() method 4 times.

But, let me also clear some more things, I have modified the code to use
redis cache and I have put the create connection block in the child_init()
method and same I see in logs. Is this the correct way to create
connections?

Thanks,
Surya



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Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

2016-02-11 Thread Alex Balashov


On 02/11/2016 08:50 PM, surya wrote:


I am facing some performance issues that's why I was thinking about those.


Performance issues of what nature?

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Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

2016-02-11 Thread surya

Thanks for the concern. I appreciate that.

It's a  long story

 
to which no one is interested.  However, I understand the community is
focused on the original product.

Basically, I modified the source for a project.( I am not working on
production Opensips). Instead of directly using Mysql, we are exposing it
using REST web service and using Redis as the cache. 

Now, after this, the performance is nowhere near to the original. I can
share more if you are still concerned :)

Thanks,
Surya



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Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

2016-02-11 Thread Alex Balashov


I'm pretty sure that only the SIP receiver threads create DB connections.

-- Alex


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Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

2016-02-11 Thread surya

Hi Alex,

Great. Got it :). Thanks.

Process::  ID=0 PID=28493 Type=attendant
Process::  ID=1 PID=28494 Type=MI FIFO
Process::  ID=2 PID=28495 Type=SIP receiver udp:10.0.1.166:5060
Process::  ID=3 PID=28496 Type=time_keeper
Process::  ID=4 PID=28497 Type=timer

Now, I understand the count.

But, Each child creates DB connections but only the PID 28495 will use.
Aren't the other connections wasted?

Thanks,
Surya



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Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

2016-02-11 Thread surya

:)

I am facing some performance issues that's why I was thinking about those.
Anyways, thanks for clearing the doubts.

Thanks,
Surya



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Re: [OpenSIPS-Users] Call-id issue in Cancel message generated by tm / $T_fr_inv_timeout

2016-02-11 Thread Bogdan-Andrei Iancu


Julian,

No need for anything more as I managed to reproduce it. Let me dig in 
and fix.


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 11.02.2016 12:07, Julian Santer wrote:

Hi Bogdan,

thank you for your time. If you need further informations (config 
files etc.) let me know.


Kind regards,
Julian Santer
Raiffeisen OnLine

Am 11.02.2016 um 10:26 schrieb Bogdan-Andrei Iancu:

Hi Julian,

I will have to test this and come back to you.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 10.02.2016 17:45, Julian Santer wrote:

Hi guys,

we seem to got the same issue like John Nash on 2015/08/12.
We use OpenSips 2.1.2 with the latest revision from git repo.

Like John we are not sure if it is a bug or our mistake ;-)

We are using topology hiding and the Call ID in the CANCEL, 
generated by the TM module, is not the same, as the call ID in the 
initial INVITE.


The call flow looks like:
PSTN carrier -> gw-carrier (topo hiding) -> core (topo hiding) -> 
gw-consumer (topo-hiding) -> UAC consumer


The CANCEL generated by the TM module of the core, sended to the 
gw-consumer is rejected by the gw-consumer.


The CANCEL starts on the core. So let me show you
1) the initial INVITE, which the core receives from the gw-carrier 
(Call-ID: GW-CARRIER)
2) the initial INVITE, which the core and sends to the gw-consumer 
(Call-ID: Core)
3) the CANCEL generated by the core after $T_fr_inv_timeout 
(Call-ID: GW-CARRIER)


1)
INVITE sip:12345@IP_CORE SIP/2.0
Via: SIP/2.0/UDP IP_GW-CARRIER:5060;branch=z9hG4bK6aa2.7710f555.0
From: ;tag=E3AE5C5C-1A42
To: 
Call-ID: 
GW-CARRIER_EjJFKHdkNlktdGM2RV93ZV5MWHdlS0wvAn1HN14LYjFHLgRiXU1aHGdCWlcE

CSeq: 101 INVITE
Max-Forwards:  8
Remote-Party-ID: 
;party=calling;screen=yes;privacy=off

Contact: 
Expires: 180
Content-Type: application/sdp
Content-Length: 474
sdp ...

2)
INVITE sip:12345@IP_UAC:PORT_UAC SIP/2.0
Via: SIP/2.0/UDP IP_CORE:5060;branch=z9hG4bK45da.82f6fd55.0
Route: 
From: ;tag=E3AE5C5C-1A42
To: 
Call-ID: 
Core_ExwGCwAcKhgvdAgnFg58LwQGAXUOXSAzC3A1JFB/FCcWCWFzGAQkXHInPFQGDzYYI3oPCCAMahZeEy11JywlCVEG

CSeq: 101 INVITE
Max-Forwards:  7
Remote-Party-ID: 
;party=calling;screen=yes;privacy=off

Contact: 
Expires: 180
Content-Type: application/sdp
Content-Length: 426
sdp ...

3)
CANCEL sip:12345@IP_UAC:PORT_UAC SIP/2.0
Via: SIP/2.0/UDP IP_CORE:5060;branch=z9hG4bK45da.82f6fd55.0
From: ;tag=E3AE5C5C-1A42
Call-ID: 
GW-CARRIER_EjJFKHdkNlktdGM2RV93ZV5MWHdlS0wvAn1HN14LYjFHLgRiXU1aHGdCWlcE

To: 
CSeq: 101 CANCEL
Max-Forwards: 70
Route: 
Reason: SIP;cause=480;text="NO_ANSWER"
User-Agent: OpenSIPS (2.1.2 (x86_64/linux))
Content-Length: 0

Kind regards,
Julian Santer
Raiffeisen OnLine


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Re: [OpenSIPS-Users] How to send a call request to every online peer?

2016-02-11 Thread Rodrigo Pimenta Carvalho

Hi Bogdan-Andrei.


Thank you very much for the hint.


By the way, I have tried it but without success.


It seems that the call request is only forwarded to the last subscriber, not 
sent to every online subscriber. See:


The code :

   append_branch();
   seturi("sip:5...@192.168.xx.yy");
   append_branch();
   seturi("sip:7...@192.168.xx.yy");
   append_branch();
   seturi("sip:6...@192.168.xx.yy");


will make the SIP INVITE be forwarded only to number 6000


And the code


   append_branch();
   seturi("sip:5...@192.168.xx.yy");
   append_branch();
   seturi("sip:6...@192.168.xx.yy");
   append_branch();
   seturi("sip:7...@192.168.xx.yy");


will make the SIP INVITE be forwarded only to number 7474. I'm trying to 
forward the SIP INVITE to 5454, 6000 and 7474 at same time.


What I'm missing? Can you see?


Any hint will be very helpful!

Thanks a lot.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



De: Bogdan-Andrei Iancu 
Enviado: terça-feira, 9 de fevereiro de 2016 09:08
Para: OpenSIPS users mailling list; Rodrigo Pimenta Carvalho
Assunto: Re: [OpenSIPS-Users] How to send a call request to every online peer?

Hi Rodrigo,

What you are looking for is called in SIP terminology "parallel forking" - a 
call attempt is sent to multiple destinations in the same time and only one can 
pick up the call.

In OpenSIPS you can easily do parallel forking by adding multiple destinations 
/ branches by using the append_branch() function:
http://www.opensips.org/Documentation/Script-CoreFunctions-2-1#toc3
[http://www.opensips.org/pub/skins/opensips/images/logo2.jpg]

openSIPS | Documentation / Core Functions - 
2.1
www.opensips.org
This section lists the all the functions exported by OpenSIPS core for script 
usage (to be used in opensips.cfg)



Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 05.02.2016 16:05, Rodrigo Pimenta Carvalho wrote:


Hi.


I have the situation:


Caller A needs call every online peer. That is, A needs call B, C, D, F...

The user should just push a 'call button' in his/her softphone and than 
OpenSIPs should receive only one SIP INVITE (let's say inviting number ). 
After receiving such INVITE, OpenSIPS should spread the request to every one. 
That is: SIP INVITE B, SIP INVITE C, SIP INVITE D, and so on.


In addiction, If one answers the call, others should receive SIP BYE 
immediately. Because A wants to call just with one peer and doesn't matter who 
will answer first. It is not a conference.


What kind of functionality am I talking about, in terms of OpenSIPS?

Is it possible?

Is the code simple?




Any hint will be very helpful!

Thanks a lot.




RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



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Re: [OpenSIPS-Users] Topology Hiding Usage Question

Re: [OpenSIPS-Users] Radius Async

Re: [OpenSIPS-Users] Call-id issue in Cancel message generated by tm / $T_fr_inv_timeout

Re: [OpenSIPS-Users] [BOOK] Building Telephony Systems with OpenSIPS - 2.1 version

Re: [OpenSIPS-Users] PJSIP example configuration not working

Re: [OpenSIPS-Users] How to TLS ?

Re: [OpenSIPS-Users] Call-id issue in Cancel message generated by tm / $T_fr_inv_timeout

Re: [OpenSIPS-Users] Call-id issue in Cancel message generated by tm / $T_fr_inv_timeout

Re: [OpenSIPS-Users] Call-id issue in Cancel message generated by tm / $T_fr_inv_timeout

Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

[OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

Re: [OpenSIPS-Users] Lots of presence:p_tm_callback, 408 warning

Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

Re: [OpenSIPS-Users] A doubt on count of child processes, children parameter in cfg.

Re: [OpenSIPS-Users] Call-id issue in Cancel message generated by tm / $T_fr_inv_timeout

Re: [OpenSIPS-Users] How to send a call request to every online peer?

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