Re: [SR-Users] Kamailio: how to route RTP media directly to endpoint

2017-08-09 Thread Mojtaba
Hi,
It is better if you want route RTP directly between UA, Dont route the
calls to Asterisk, And do this like below:
UAC1---sip>kamailio->UAC2
In asterisk, there is directmedia options for handle RTP.
Be notice you should use STUN in this regards. becuase of type of nats
in clients, you have some challenge.

On Wed, Aug 9, 2017 at 6:18 PM, wsotest.512 <wsotest@gmail.com> wrote:
> Hi all,
>
> We have usual config Kamailio + Asterisk where Kamailio play as sip and rtp
> proxy. Kamailio have public IP, asterisk – no. All calls between clients now
> going like that:
>
>
>
> UserA ---sip--> Kamailio --> Asterisk --> UserB
>
>\-rtp--> Kamailio (rtpproxy) --> Asterisk --> UserB
>
>
>
> All clients of course from Internet and behind Nat. Main problem is amount
> of traffic going through Kamailio and Asterisk. We need to pay for every
> additional GB behind limit in tariff plan to hosting provider.
>
> So we decided to try route all rtp traffic between users directly.
>
>
>
> UserA ---sip--> Kamailio --> Asterisk --> UserB
>
>\-rtp--> --> --> UserB
>
>
>
> Is it possible at all? Maybe someone already did it …
>
>
>
>
>
> --
>
> BR, Alex
>
>
>
>
>
>
> _______
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> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>



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Re: [SR-Users] Setup IMS core Network When Network behind nat

2017-07-11 Thread Mojtaba
Hi.
I found where was my problem. I had a misunderestanding. I used public
domaim as outbound proxy and i could register to ims.
On 11 Jul 2017 15:00, "Mojtaba" <mes...@gmail.com> wrote:

> Hi Daniel.
> Let me describe more clearly the network. Let me suppose these
> configurations:
> Private IMS Domain: kamailio0ims.test (192.168.100.200)
> username: b...@kamailio-ims.test
> secret: bob
> outboundproxy: pcscf.kamailio-ims.test (192.168.100.101)
> With these informations, all things works correctly.
> Now suppose we have one public IP with predefined
> domain(sbc.ims-domain.com) that is set on PCSCF.(This node has two ip
> address, one private ip and on public ip),
> How can the bob register to network like as before?because the domain
> sbc.ims-domain.com is not validate on IMS network.
> should i use the sems-sbc? becuase in this situation the private
> domain is not reachables.
> If it is needed, let me to describe more.
> Thanks
>
>
>
> On Tue, Jul 11, 2017 at 2:24 PM, Daniel-Constantin Mierla
> <mico...@gmail.com> wrote:
> > Hello,
> >
> > do you want the entire platform to be behind a single public ip, or each
> > node in the network to run on a private ip but have also a public ip
> > associated?
> >
> > Cheers,
> > Daniel
> >
> >
> > On 11.07.17 11:31, Mojtaba wrote:
> >> Hello, I am working on IMS core network. In this network we have these
> >> node with private IP:
> >> P-CSCF: 192.168.100.101(pcscf.kamailio-ims.test)
> >> I-CSCF: 192.168.100.102(icscf.kamailio-ims.test)
> >> S-CSCF: 192.168.100.103(scscf.kamailio-ims.test)
> >> HSS: 192.168.100.200(hss.kamailio-ims.test)
> >> I have two default users (bob and alice) and all things right great.
> >> Now i want use this network with one public IP behind nat. Let me know
> >> What should i do?
> >> In one way i could use SEMS-SBC and set public IP on it, How should i
> >> config sems-sbc to work correctly?
> >> Thanks.With Regads.
> >> M.ES.S
> >>
> >> ___
> >> Kamailio (SER) - Users Mailing List
> >> sr-users@lists.kamailio.org
> >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
> >
> > --
> > Daniel-Constantin Mierla
> > www.twitter.com/miconda -- www.linkedin.com/in/miconda
> > Kamailio Advanced Training - www.asipto.com
> > Kamailio World Conference - www.kamailioworld.com
> >
>
>
>
> --
> --Mojtaba Esfandiari.S
>
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[SR-Users] Setup IMS core Network When Network behind nat

2017-07-11 Thread Mojtaba
Hello, I am working on IMS core network. In this network we have these
node with private IP:
P-CSCF: 192.168.100.101(pcscf.kamailio-ims.test)
I-CSCF: 192.168.100.102(icscf.kamailio-ims.test)
S-CSCF: 192.168.100.103(scscf.kamailio-ims.test)
HSS: 192.168.100.200(hss.kamailio-ims.test)
I have two default users (bob and alice) and all things right great.
Now i want use this network with one public IP behind nat. Let me know
What should i do?
In one way i could use SEMS-SBC and set public IP on it, How should i
config sems-sbc to work correctly?
Thanks.With Regads.
M.ES.S

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[SR-Users] I_SCSCF_select function dont return the next SCSCF in IMS network!!!

2017-07-23 Thread Mojtaba
Hello,
Today i have a strange problem in IMS network.
All nodes (pcscf,icscf,scscf and hss) are configured in independent server.
My domain is default(open-ims.test)

Now when i want to register b...@open-ims.test, I get this error from ICSCF:
500 Error in request.
When i saw the log files, i found the icscf could not found the next
node(scscf) in registration path. As i defined prefered scscf_uri in
configuration file, Why the icscf, could not found the scscf?
The dns server works correctly and i have ping of scscf.open-ims.test
from icscf node.
When i trace the diameter packets, the icscf connect to hss
successfully but dont send UAR to hss while registration proccess!!
It's strange.
Please let me know why i have this problem,
Thanks With Regards.
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Re: [SR-Users] I_SCSCF_select function dont return the next SCSCF in IMS network!!!

2017-07-24 Thread Mojtaba
Hi,
I had some search about this issue, I found the same issue that why
the ICSCF doesn't send UAR to HSS, I think this issue is fixed with
commit 987c810 
(https://github.com/kamailio/kamailio/commit/987c8101504cc7939c6782df55d7073f492e12c8).
I hope i could solved this problem :))

On Sun, Jul 23, 2017 at 3:38 PM, Mojtaba <mes...@gmail.com> wrote:
> Hello,
> Today i have a strange problem in IMS network.
> All nodes (pcscf,icscf,scscf and hss) are configured in independent server.
> My domain is default(open-ims.test)
>
> Now when i want to register b...@open-ims.test, I get this error from ICSCF:
> 500 Error in request.
> When i saw the log files, i found the icscf could not found the next
> node(scscf) in registration path. As i defined prefered scscf_uri in
> configuration file, Why the icscf, could not found the scscf?
> The dns server works correctly and i have ping of scscf.open-ims.test
> from icscf node.
> When i trace the diameter packets, the icscf connect to hss
> successfully but dont send UAR to hss while registration proccess!!
> It's strange.
> Please let me know why i have this problem,
> Thanks With Regards.
> --
> --Mojtaba Esfandiari.S



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Re: [SR-Users] I_SCSCF_select function dont return the next SCSCF in IMS network!!!

2017-07-24 Thread Mojtaba
Hi again.
I fixed the problem with the commit 987c810
(https://github.com/kamailio/kamailio/commit/987c8101504cc7939c6782df55d7073f492e12c8).
in kamailio-5.0.1
In kamailio-5.0.2 it is fixed,
Thanks Mr Carsten Bock & ng_voice

On Mon, Jul 24, 2017 at 11:35 AM, Mojtaba <mes...@gmail.com> wrote:
> Hi,
> I had some search about this issue, I found the same issue that why
> the ICSCF doesn't send UAR to HSS, I think this issue is fixed with
> commit 987c810 
> (https://github.com/kamailio/kamailio/commit/987c8101504cc7939c6782df55d7073f492e12c8).
> I hope i could solved this problem :))
>
> On Sun, Jul 23, 2017 at 3:38 PM, Mojtaba <mes...@gmail.com> wrote:
>> Hello,
>> Today i have a strange problem in IMS network.
>> All nodes (pcscf,icscf,scscf and hss) are configured in independent server.
>> My domain is default(open-ims.test)
>>
>> Now when i want to register b...@open-ims.test, I get this error from ICSCF:
>> 500 Error in request.
>> When i saw the log files, i found the icscf could not found the next
>> node(scscf) in registration path. As i defined prefered scscf_uri in
>> configuration file, Why the icscf, could not found the scscf?
>> The dns server works correctly and i have ping of scscf.open-ims.test
>> from icscf node.
>> When i trace the diameter packets, the icscf connect to hss
>> successfully but dont send UAR to hss while registration proccess!!
>> It's strange.
>> Please let me know why i have this problem,
>> Thanks With Regards.
>> --
>> --Mojtaba Esfandiari.S
>
>
>
> --
> --Mojtaba Esfandiari.S



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Re: [SR-Users] IOS 10 and SIP

2017-06-30 Thread Mojtaba
Hi.
The TSILO  module in kamailio is impelemented for push notifications for
ISO  and Android.
Please read its documentations in kamailio website.
With Regards
Mojtaba Esfandiari.S
On 30 Jun 2017 16:37, "Kjeld Flarup" <k...@viptel.dk> wrote:

> Rumours is that Apple no longer accepts apps which can do persistent
> connections in the background.
>
> To my best knowledge that means that IOS no longer supports SIP incoming
> calls.
> The app should now use Push Notifications, but SIP does not support this.
>
> Anybody faced this problem?
>
> Is the solution to call an external program when processing an Invite?
> This leaves some issues.
>
> 1. Should we wait some seconds before proceeding or wait until the Push
> notification has been processed, to give the app time to register.
> We cannot start to send invites, before we have a correct registration. An
> old registration may change ports when the app wakes up and makes a new
> register via NAT.
>
> 2. How do we know, if a given user should have a push notification?
>
> 3. Which information do we need to be able to send the push notification?
>
> 4. Does this give a penalty to Android users, because we have to wait for
> IOS?
>
>
>
> --
> Med venlig hilsen / Best regards
> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
> Viptel ApS, Hammershusvej 16C, DK-7400 Herning
> Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
>
>
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Re: [SR-Users] remote ip address of sender and receiver

2017-06-30 Thread Mojtaba
Let me know how do you test it?
You should write it in Request-route block
On 30 Jun 2017 13:04, "vuleetu" <vule...@gmail.com> wrote:

> I tried, but no luck. Could you tell which route block i should place
> these code to?
>
> On Thu, Jun 29, 2017 at 10:57 PM, Mojtaba <mes...@gmail.com> wrote:
>
>> Hi.
>> I wrote this code, I check do the both peers are in the same network
>> or not. It works correctly.
>> # Check if both peers are in the same network(nat).
>> if (is_method("INVITE")){
>>  lookup("location","m");
>>  $var(x) = $(ru{s.select,1,@});
>>  $var(y) = $(var(x){s.select,0,:});
>>  if ($var(y) == $si)
>> $var(nat) = "yes";
>>  else
>> $var(nat) = "no";
>>
>> }
>> With Regagrds.
>> Mojtaba Esfandiari.S
>>
>> On Thu, Jun 29, 2017 at 6:57 PM, vuleetu <vule...@gmail.com> wrote:
>> > Hi, all
>> >
>> >Is there any way to get remote ip address of both sender($si) and
>> > receiver(?)? I am configuring kamailio with multiple rtpproxy sets.
>> > Basically, rtpproxy set will be choosed based on the country those ip
>> > addresses of caller and callee belong to. Lets say if caller and callee
>> are
>> > both in US, set 1 will selected. If they are not in the same country,
>> set 2
>> > will be selected.  I dont know if there is any way to do that.
>> >
>> >
>> > Best regards
>> >
>> > Fisher
>> >
>> > ___
>> > Kamailio (SER) - Users Mailing List
>> > sr-users@lists.kamailio.org
>> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>> >
>>
>>
>>
>> --
>> --Mojtaba Esfandiari.S
>>
>> ___
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>>
>
>
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Re: [SR-Users] IOS 10 and SIP

2017-06-30 Thread Mojtaba
Hi Federico.
I saw your speech at Kamailio world conferance. That's greats. :))
With Regards.
Mojtaba Esfandiari.S
On 30 Jun 2017 17:02, "Federico Cabiddu" <federico.cabi...@gmail.com> wrote:

> Hi,
> good news: you can easily handle this scenario with Kamailio!
> If you want to have an overall view of VoIP you can have a
> look at this speech I gave at Kamailio World 2015:
> https://www.youtube.com/watch?v=4XIrR9bwUkM.
> And the slides: https://www.kamailio.org/events/2015-KamailioWorld/D
> ay2/20-Federico.Cabiddu-Kamailio-In-A-Mobile-World.pdf.
>
> Cheers,
>
> Federico
>
> On Fri, Jun 30, 2017 at 2:07 PM, Kjeld Flarup <k...@viptel.dk> wrote:
>
>> Rumours is that Apple no longer accepts apps which can do persistent
>> connections in the background.
>>
>> To my best knowledge that means that IOS no longer supports SIP incoming
>> calls.
>> The app should now use Push Notifications, but SIP does not support this.
>>
>> Anybody faced this problem?
>>
>> Is the solution to call an external program when processing an Invite?
>> This leaves some issues.
>>
>> 1. Should we wait some seconds before proceeding or wait until the Push
>> notification has been processed, to give the app time to register.
>> We cannot start to send invites, before we have a correct registration.
>> An old registration may change ports when the app wakes up and makes a new
>> register via NAT.
>>
>> 2. How do we know, if a given user should have a push notification?
>>
>> 3. Which information do we need to be able to send the push notification?
>>
>> 4. Does this give a penalty to Android users, because we have to wait for
>> IOS?
>>
>>
>>
>> --
>> Med venlig hilsen / Best regards
>> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
>> Viptel ApS, Hammershusvej 16C, DK-7400 Herning
>> Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
>>
>>
>> ___
>> Kamailio (SER) - Users Mailing List
>> sr-users@lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
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Re: [SR-Users] remote ip address of sender and receiver

2017-06-29 Thread Mojtaba
Hi.
I wrote this code, I check do the both peers are in the same network
or not. It works correctly.
# Check if both peers are in the same network(nat).
if (is_method("INVITE")){
 lookup("location","m");
 $var(x) = $(ru{s.select,1,@});
 $var(y) = $(var(x){s.select,0,:});
 if ($var(y) == $si)
$var(nat) = "yes";
 else
$var(nat) = "no";

}
With Regagrds.
Mojtaba Esfandiari.S

On Thu, Jun 29, 2017 at 6:57 PM, vuleetu <vule...@gmail.com> wrote:
> Hi, all
>
>Is there any way to get remote ip address of both sender($si) and
> receiver(?)? I am configuring kamailio with multiple rtpproxy sets.
> Basically, rtpproxy set will be choosed based on the country those ip
> addresses of caller and callee belong to. Lets say if caller and callee are
> both in US, set 1 will selected. If they are not in the same country, set 2
> will be selected.  I dont know if there is any way to do that.
>
>
> Best regards
>
> Fisher
>
> ___
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> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>



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Re: [SR-Users] Kamailio IMS NAT issue: PRACK/ACK RURI changing in P-CSCF

2017-06-21 Thread Mojtaba
Hi,
You mean that we should use either SBC or Proxy-CSCF (don't both of
them) as entry point of IMS network? Is it right?
Thank.
Mojtaba Esfandiari.S


On Wed, Jun 21, 2017 at 12:43 PM, Carsten Bock <cars...@ng-voice.com> wrote:
> Hi,
>
> There is a common scenario, where Kamailio needs to be in front of e.g SEMS:
> PCRF Integration for QoS. You need to access the IP of the sender and the
> SDP.
> You would have to integrate that into SEMS, but if you do, you can actually
> drop Kamailio as a Proxy-CSCf completely...
>
> Thanks,
> Carsten
>
> Am 21.06.2017 09:48 schrieb "Mojtaba" <mes...@gmail.com>:
>>
>> Is it not better to use SEMS as SBC in the front of your network? And
>> all P/S/I/CSCF have the private IP address.
>> Thanks
>> --Mojtaba Esfandiari.S
>>
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Re: [SR-Users] Kamailio IMS NAT issue: PRACK/ACK RURI changing in P-CSCF

2017-06-22 Thread Mojtaba
Hi
Thank you for your explanation.
What is your suggestions about the setup IMS Core Network for area? I
mean which of the senario is better?
1) All nodes (P/I/S-CSCF and HSS) have public IP Address in IMS Core
Network, While the subscriber just could see PCSCF. In this way we may
use some modules to hide topology, e.g TOPOS module.
2) We have one Public IP Address and one Private IP Address for PCSCF,
While other nodes (I/S-CSCF and HSS) have Private IP Address. In this
way we use Nat modules (or RTP Proxy) for relay RTP. One of major
functionality of nat module is relaying RTP between Public IP address
and Private IP Address in PCSCF node.
Thanks.

Mojtaba Esfandiari.S

On Thu, Jun 22, 2017 at 12:28 AM, Carsten Bock <cars...@ng-voice.com> wrote:
> Hi,
>
> you can for example buy a Proxy-CSCF from Oracle (the former
> AcmePacket NET-NET SBC's), which provides such functionality.
>
> I'm working on a new Version of the config, which supports some more
> options here:
> 1) Use topos, if e.g. no transcoding is needed
> 2) Use SEMS or Asterisk like before (with some cleanups/optimizations)
>
> When I talk with customers about commercial deployments, they often
> refer to Kamailio as a SBC (and since there is no real definition for
> a SBC, that's probably valid); since it does most of the things, a SBC
> would do (topology hiding, DOS-Attack detection, Limiting,
> Header-Manilpulation, Validating of the messages, ...).
> For me one of the top-reasons for using either SEMS, Frafos ABC-SBC or
> Asterisk is Transcoding & Topology hiding, as almost everything else
> can be done on Kamailio too.
>
> Thanks,
> Carsten
>
>
>
>
> 2017-06-21 13:15 GMT+02:00 Mojtaba <mes...@gmail.com>:
>> Hi,
>> You mean that we should use either SBC or Proxy-CSCF (don't both of
>> them) as entry point of IMS network? Is it right?
>> Thank.
>> Mojtaba Esfandiari.S
>>
>>
>> On Wed, Jun 21, 2017 at 12:43 PM, Carsten Bock <cars...@ng-voice.com> wrote:
>>> Hi,
>>>
>>> There is a common scenario, where Kamailio needs to be in front of e.g SEMS:
>>> PCRF Integration for QoS. You need to access the IP of the sender and the
>>> SDP.
>>> You would have to integrate that into SEMS, but if you do, you can actually
>>> drop Kamailio as a Proxy-CSCf completely...
>>>
>>> Thanks,
>>> Carsten
>>>
>>> Am 21.06.2017 09:48 schrieb "Mojtaba" <mes...@gmail.com>:
>>>>
>>>> Is it not better to use SEMS as SBC in the front of your network? And
>>>> all P/S/I/CSCF have the private IP address.
>>>> Thanks
>>>> --Mojtaba Esfandiari.S
>>>>
>>>> ___
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>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
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>>>
>>
>>
>>
>> --
>> --Mojtaba Esfandiari.S
>>
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>
>
>
> --
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> CEO (Geschäftsführer)
>
> ng-voice GmbH
> Millerntorplatz 1
> 20359 Hamburg / Germany
>
> http://www.ng-voice.com
> mailto:cars...@ng-voice.com
>
> Office +49 40 5247593-40
> Fax +49 40 5247593-99
>
> Sitz der Gesellschaft: Hamburg
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>
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[SR-Users] Strange issue in registering users in Kamailio

2017-06-11 Thread Mojtaba
Hello, In now days i have a strange issue in registering users in kamailio.
My senario is the following:
I setup kamailio with basic configurations and i created 2000 users in
mysql(from 1 to12000). With registration senario with SIPP tool, i
have not issue and every things works right,(Notice that we know SIPP
uses the same source ip for all registration), But when i use Spirent
Abacus 5000(Test solution for voice, VoLTE and IMS), the kamailio just
register 1024 users and other users could not register to kamailio.
Spirent Abacus 5000 use the different source ip for each registration.
I have these error in syslog message:
ERROR: sl [../../core/forward.h:201]: msg_send buffer(): udp_send failed
one possible reason is the server is bound to localhost
I set this parameter in kamailio configurations, but the issue dont solved:
open_files_limits:2048
and in linux configuration:
ulimit -n 99
Please let me know how can i fixed it?

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Re: [SR-Users] Maximum Registration on Kamailo??

2017-09-13 Thread Mojtaba
Hi,
This restriction is not regarded to kamailio.
It is from arp table  in your linux. The default rows of the arp table is
1024. you should increase it more than 1024 rows.
With Regards.Mojtaba Esfanfiari.S
On 13 Sep 2017 23:05, "amit Kumar"  wrote:

> Hi All,
>
> kamctl ul show => show below result, does that mean max 1024 contact can
> be registered, if that is the case how to increase the max registration
> limit.
>
> Domain:: location table=*1024 *records=2 max_slot=1
>
> Thanks
>
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Re: [SR-Users] telcosuite3-crm sql

2017-09-05 Thread Mojtaba
Hi,
Would you please share it so that i have it too?
With Regards.Mojtaba.ES


On Thu, Aug 31, 2017 at 11:09 AM, Carsten Bock <cars...@ng-voice.com> wrote:
> Hi,
>
> you find all the SQL files in /var/www/management and ./rest, as you
> need the REST-API for the Management-Portal.
>
> Thanks,
> Carsten
>
> 2017-08-31 8:17 GMT+02:00 Kordován Szabolcs <kos...@gmail.com>:
>> Hi,
>>
>> I would like to try the IMS according to the Kamailio World 2017: Kamailio
>> For IMS And VoLTE presentation.
>> Unfortunately I have problem with telcosuite, the sql structure is missing.
>> Where can I find the whole create.sql?
>>
>> Thanks for help.
>> Regards,
>> Szabolcs
>>
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>
>
>
> --
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> CEO (Geschäftsführer)
>
> ng-voice GmbH
> Millerntorplatz 1
> 20359 Hamburg / Germany
>
> http://www.ng-voice.com
> mailto:cars...@ng-voice.com
>
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Re: [SR-Users] [Kamailio-Business] Astricon 2017

2017-09-25 Thread Mojtaba
Hello Mr Carsten Bock and Mr Danial Constantin
I wish i could be there,
I hope i could be on Kamailio Conferance in next year.
Thanks
On 25 Sep 2017 11:10, "Carsten Bock"  wrote:

> Hi,
>
> just a note:
> I give a talk on how Asterisk fits into an Kamailio IMS deployment: On
> thursday at 10 AM, don't be late! :-)
>
> I'll be there, too!
>
> See you there,
> Carsten
>
>
>
> 2017-09-25 10:04 GMT+02:00 Daniel-Constantin Mierla :
> > Hello,
> >
> > Kamailio will be present at Astricon 2017 having a stand in the expo
> > area (Orlando, FL, during October 3-5 -- more details at
> www.astricon.net).
> >
> > Should you participate at the event and have flyers/other materials with
> > products/solutions/services that use Kamailio or giveaways, you are
> > welcome to join the booth or just drop them there, we are more than
> > happy to promote and spread them. Also, you are very welcome to just
> > stop by and have a chat.
> >
> > Besides the activity in the exhibition area, I and Fred Posner will have
> > presentations about using Kamailio together with Asterisk or other VoIP
> > applications. Be sure you do not miss them, the agenda is available at:
> >
> >   - http://www.asterisk.org/community/astricon-user-conference/schedule
> >
> > Expect a consistent group from Kamailio community, it's a great event to
> > be at!
> >
> > Looking forward to meeting many of you next week in Orlando!
> >
> > Cheers,
> > Daniel
> >
> > --
> > Daniel-Constantin Mierla
> > www.twitter.com/miconda -- www.linkedin.com/in/miconda
> > Kamailio Advanced Training - www.asipto.com
> > Kamailio World Conference - www.kamailioworld.com
> >
> >
> > ___
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> > http://lists.openser-project.org/cgi-bin/mailman/listinfo/business
>
>
>
> --
> Carsten Bock
> CEO (Geschäftsführer)
>
> ng-voice GmbH
> Millerntorplatz 1
> 20359 Hamburg / Germany
>
> http://www.ng-voice.com
> mailto:cars...@ng-voice.com
>
> Office +49 40 5247593-40
> Fax +49 40 5247593-99
>
> Sitz der Gesellschaft: Hamburg
> Registergericht: Amtsgericht Hamburg, HRB 120189
> Geschäftsführer: Carsten Bock
> Ust-ID: DE279344284
>
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Re: [SR-Users] "Federated" PBX solution with GUI to use with Kamailio - Suggestions needed

2017-11-22 Thread Mojtaba
Hello,
I think the Kazoo form 2600Hz  group is the good choice.
Although that solution is for carrier and cluster base, But i think
you could use it in your solutions.
Another my choice in this regard is Simotel PBX, It is asterisk base
sulotios (without Kamailio, Non multi-tenant).
With Regards.
Mojtaba

On Thu, Nov 23, 2017 at 7:41 AM, George Diamantopoulos
<georged...@gmail.com> wrote:
> Hello all!
>
> I'm looking at "PBX" platforms to use with kamailio as a load-balancer. This
> is not a strictly kamailio question, but I thought that if anyone has a
> suggestion to make, they would be watching this list.
>
> I'm investigating a use case with the following requirements:
> * Kamailio will act as a load-balancer, dispatching initial requests to PBX
> entities (e.g. Asterisk, Freeswitch)
> * PBX solution must have a single GUI interface to manage PBX functions
> (think FreePBX, FusionPBX etc)
> * Configuration via the GUI must apply to all PBX instances
> * Some sort of "asterisk-realtime + device state synchronization"-like setup
> must ensure every PBX box can service any request at any time (for failover)
> * Solution needn't be multi-tenant oriented, but if it is, it must work fine
> for single-tenant setups as well without multi-tenant capability getting in
> the way all the time
> * Feature list needn't be as full as, say, FreePBX's, but most common
> functionality must be available (inbound/outbound routing, trunks, queues,
> hunting to name a few)
> * Must be open source
>
> As a proprietary example, MiRTA PBX seems to check all the boxes above,
> apart from being open source. Now this "product" also offers high
> availability for databases and web apps, which is beyond the scope of this
> investigation. I'm only looking at a solution that provides a federated,
> synchronized farm of PBX instances.
>
> Any input would be greatly appreciated. Thanks!
>
> Best regards,
> George
>
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Re: [SR-Users] max_contacts does not work as expected

2017-11-16 Thread Mojtaba
Hello.
In Kamailio when the uac is registerd in new AOR, the other AOR is take
afew minute until expired time is occure.
As you mentioned you set db_mode=3, becuase the database is should be
updated,It may take more time to update.
Be notice that if the two different AOR is exist in kamailio,No more
Rrgister requests is accepted with Kamailio.
With Regards.
Mojtaba Esfandiari.S
On 16 Nov 2017 19:22, "Gholamreza Sabery" <gr.sab...@gmail.com> wrote:

> No ideas?
>
> On Wed, Nov 15, 2017 at 3:27 PM, Gholamreza Sabery <gr.sab...@gmail.com>
> wrote:
>
>> Hello Daniel,
>>
>> I use a value of 3 for db_mode.
>>
>> On Wed, Nov 15, 2017 at 11:11 AM, Daniel-Constantin Mierla <
>> mico...@gmail.com> wrote:
>>
>>> Hello,
>>>
>>> what is the value of db_mode parameter for usrloc module?
>>>
>>> Cheers,
>>> Daniel
>>>
>>> On 15.11.17 06:36, Gholamreza Sabery wrote:
>>>
>>> I have a Kamailio server that has:
>>>
>>> modparam("registrar", "max_contacts", 2)
>>>
>>> But recently I noticed some users have more than two records in the
>>> location table. For example one of the users has three different records
>>> with nearly the same information. The records are as follows. Is this
>>> behavior normal? (I am using Kamailio 4XX).
>>>
>>> *** 1. row ***
>>>id: 171296745
>>>  ruid: uloc-2c69-59db6c73-7ca-132d92
>>>  username: 
>>>domain: NULL
>>>   contact: sip:@5.106.35.18:10362
>>>  received: sip:5.106.35.18:10361
>>>  path: NULL
>>>   expires: 2017-11-14 08:45:50
>>> q: -1.00
>>>callid: XpCnmaAhRF
>>>  cseq: 43
>>> last_modified: 2017-11-14 08:44:50
>>> flags: 0
>>>cflags: 64
>>>user_agent: android_1.1.0_samsung_SM-E500H
>>> _7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2)
>>>socket: udp:192.168.103.21:5060
>>>   methods: NULL
>>>  instance: 
>>>reg_id: 0
>>> server_id: 11369
>>> connection_id: -1
>>> keepalive: 1
>>> partition: 14
>>> *** 2. row ***
>>>id: 171296746
>>>  ruid: uloc-2c69-59db6c73-7f2-eb5d92
>>>  username: 
>>>domain: NULL
>>>   contact: sip:@5.106.35.18:10362
>>>  received: sip:5.106.35.18:10361
>>>  path: NULL
>>>   expires: 2017-11-14 08:45:50
>>> q: -1.00
>>>callid: BzMKuyJN8C
>>>  cseq: 50
>>> last_modified: 2017-11-14 08:44:50
>>> flags: 0
>>>cflags: 64
>>>user_agent: android_1.1.0_samsung_SM-E500H
>>> _7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2)
>>>socket: udp:192.168.103.21:5060
>>>   methods: NULL
>>>  instance: 
>>>reg_id: 0
>>> server_id: 11369
>>> connection_id: -1
>>> keepalive: 1
>>> partition: 23
>>> *** 3. row ***
>>>id: 171296747
>>>  ruid: uloc-2c69-59db6c73-80f-541d92
>>>  username: 
>>>domain: NULL
>>>   contact: sip:@5.106.35.18:10362
>>>  received: sip:5.106.35.18:10361
>>>  path: NULL
>>>   expires: 2017-11-14 08:45:50
>>> q: -1.00
>>>callid: 3p9P4PgZSv
>>>  cseq: 51
>>> last_modified: 2017-11-14 08:44:50
>>> flags: 0
>>>cflags: 64
>>>user_agent: android_1.1.0_samsung_SM-E500H
>>> _7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2)
>>>socket: udp:192.168.103.21:5060
>>>   methods: NULL
>>>  instance: 
>>>reg_id: 0
>>> server_id: 11369
>>> connection_id: -1
>>> keepalive: 1
>>> partition: 15
>>>
>>>
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>>>
>>>
>>> --
>>> Daniel-Constantin Mierlawww.twitter.com/miconda -- 
>>> www.linkedin.com/in/miconda
>>> Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com
>>> Kamailio World Conference - www.kamailioworld.com
>>>
>>>
>>
>
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Re: [SR-Users] Kamailio IMS + Asterisk PBX

2017-11-21 Thread Mojtaba
Hello,
Maybe this link be useful to understanding the concept of Asterisk
working as Telephone Application Servers(TAS) in IMS core network.
https://www.youtube.com/watch?v=3nVRpzkUbIg
With Regards. Mojtaba

On Mon, Nov 20, 2017 at 5:14 PM, Amar Tinawi <amar.tin...@gmail.com> wrote:
> Dears;
> Good day !
>
> is it possible to use Asterisk pbx server as an Application server for the
> kamailio IMS users ? as if i want to correlate an IMS user with an extension
> number.
>
> My though is kamailio ims  will handle the registration process  and
> authentication and Asterisk will deal with IMS users as they are PBX users
> to get all asterisk's benefits (call conference, monitoring, waiting, voice
> mail ... etc)
>
> i'm thinking to integrate Asterisk with S-cscf, and when an IMS user
> register to the IMS to domain, a trigger will initiate the S-cscf to send
> notify to asterisk .
>
>
>
> i didnt find any solution or tutorial for this procedure, and idont know if
> it is possible to be done.
>
> Regards
> Ammar
>
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Re: [SR-Users] Kamailio IMS + Asterisk PBX

2017-11-21 Thread Mojtaba
Hello,
Maybe this link be useful to understanding the concept of Asterisk
working as Telephone Application Servers(TAS) in IMS core network.
https://www.youtube.com/watch?v=3nVRpzkUbIg
With Regards. Mojtaba

On Tue, Nov 21, 2017 at 11:13 AM, Daniel-Constantin Mierla
<mico...@gmail.com> wrote:
> Hello,
>
>
> On 20.11.17 14:44, Amar Tinawi wrote:
>> Dears;
>> Good day !
>>
>> is it possible to use Asterisk pbx server as an Application server for
>> the kamailio IMS users ? as if i want to correlate an IMS user with an
>> extension number.
>>
>> My though is kamailio ims  will handle the registration process  and
>> authentication and Asterisk will deal with IMS users as they are PBX
>> users to get all asterisk's benefits (call conference, monitoring,
>> waiting, voice mail ... etc)
>>
>> i'm thinking to integrate Asterisk with S-cscf, and when an IMS user
>> register to the IMS to domain, a trigger will initiate the S-cscf to
>> send notify to asterisk .
>>
>> i didnt find any solution or tutorial for this procedure, and idont
>> know if it is possible to be done.
>>
> not sure if IMS has any relevant requirement here, but using Kamailio
> with Asterisk is very common. There are several options to integrate the
> two, for some of them I wrote tutorials available at:
>
>   - https://kb.asipto.com/asterisk:index
>
> Cheers,
> Daniel
>
> --
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> www.twitter.com/miconda -- www.linkedin.com/in/miconda
> Kamailio Advanced Training - www.asipto.com
> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
>
>
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Re: [SR-Users] STUN

2018-05-07 Thread Mojtaba
Hello,
You could run STUN server on the same server that Kamailio is running,
It is very straightforward.
Just donload STUNTMAN ,An open source STUN server - Version 1.2.7, and
read README file.

Alternative, Opensips has build-in STUN module, You could use it,too
With Regards.Mojtaba

On Sun, May 6, 2018 at 8:18 PM, Sergiu Pojoga <pojo...@gmail.com> wrote:
> Simple, load the stun module then point the device' stun address to your
> kamailio host on port 5060 (or whatever SIP listening port is configured)
>
> On Sun, May 6, 2018 at 11:12 AM, Wilkins, Steve <swwilk...@mitre.org> wrote:
>>
>> Hello all,
>>
>>
>>
>> Does anyone know the current status of how to enable STUN.  I am using
>> Kamailio Ver 5.x.
>>
>>
>>
>> Thank you,
>>
>> -Steve
>>
>>
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>
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Re: [SR-Users] STUN

2018-05-07 Thread Mojtaba
The STUN solutions for NAT traversal is classified as near-end.This
solutions implemented on the client-side, Then you don't need to
configure kamailio.
Be notice only the TURN solution is used on server-side. The TURN
solution  needs some configurations in Kamailio.
With Regards.Mojtaba

On Mon, May 7, 2018 at 5:29 PM, Wilkins, Steve <swwilk...@mitre.org> wrote:
> So If I already have a Stun Server, I don’t need to configure Kamailio to
> use that Server…Is that correct?
>
>
>
> Thank you
>
>
>
> From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of
> Sergiu Pojoga
> Sent: Sunday, May 6, 2018 11:49 AM
> To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
> Subject: Re: [SR-Users] STUN
>
>
>
> Simple, load the stun module then point the device' stun address to your
> kamailio host on port 5060 (or whatever SIP listening port is configured)
>
>
>
> On Sun, May 6, 2018 at 11:12 AM, Wilkins, Steve <swwilk...@mitre.org> wrote:
>
> Hello all,
>
>
>
> Does anyone know the current status of how to enable STUN.  I am using
> Kamailio Ver 5.x.
>
>
>
> Thank you,
>
> -Steve
>
>
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Re: [SR-Users] Updated NAT traversal article

2018-05-12 Thread Mojtaba
Hi,
That's greats concept in this regards.
I think it could be great to describe about SEMS (e.g B2BUA,
NAT-Traversal) and working it with Kamailio.
The SEMS has proper modules to solve NAT Traversal in Kamailio,too
If you want, I could give you it's documents.
Thanks With Regards.Mojtaba

On Fri, May 11, 2018 at 6:43 PM, Alex Balashov
<abalas...@evaristesys.com> wrote:
> Hi,
>
> I have updated this article with some other topics:
>
> http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-definitive-guide/
>
> "The definitive guide" is of course an ambitious and moving target, so
> there may be more evolution in the future.
>
> -- Alex
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
>
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
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Re: [SR-Users] ASN.1 HI2- IRI-Parameters

2018-06-06 Thread Mojtaba
Hi Daniel
Finally i could implement 3GPP TS 101-671 (HI2Operations) in C
programming with asn1c.
I could use it with Kamailio like Homer.
Daniel is it could be a new module in Kamailio? How can i implement it
as a new module in Kamailio? I want to do it.
Thanks.
Mojtaba

On Mon, Apr 30, 2018 at 2:00 PM, Mojtaba  wrote:
> Any idea in this regards?
> I understood it's structure of HI2Operations as ASN.1 IRI-HI2, I attached it.
> I think i should convert SIP signalling flow to this structure for 
> transmission.
> Is it right way that i am doing?
> With Regards.Mojtaba
>
> On Thu, Apr 26, 2018 at 12:50 PM, Mojtaba  wrote:
>> Hello,
>> The declaration of details are in 3GPP TS 101-671 Annex D.5 (ASN.1
>> description of IRI (HI2 interface)).
>> I need a sample packet of HI2Operations for underestanding, so that
>> i'll develop this HI in kamailio.
>> With Regards.Mojtaba
>>
>>
>> On Thu, Apr 26, 2018 at 11:20 AM, Daniel-Constantin Mierla
>>  wrote:
>>> Hello,
>>>
>>> can you provide more details or references (web links) about the operations
>>> you want to do? Like what sip message comes to kamailio and what you need to
>>> change to it.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>> On 25.04.18 06:00, Mojtaba wrote:
>>>
>>> Hello.
>>> I need to use HI2Operations for transfering IRI-Parameters in kamailio.Dose
>>> anybody have experience in this regards?
>>> Thanks.
>>> Mojtaba
>>>
>>>
>>> ___
>>> Kamailio (SER) - Users Mailing List
>>> sr-users@lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>> --
>>> Daniel-Constantin Mierla
>>> www.twitter.com/miconda -- www.linkedin.com/in/miconda
>>> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
>>
>>
>>
>> --
>> --Mojtaba Esfandiari.S
>
>
>
> --
> --Mojtaba Esfandiari.S



-- 
--Mojtaba Esfandiari.S

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Re: [SR-Users] RTPEngine Recording does not work

2018-07-01 Thread Mojtaba
I have strange issue,
the output of of `uname -r` command is "3.16.0-4-amd64", while all the
debs file are "4.9.0-0.bpo.6-amd64".
ngcp-rtpengine_6.4.0.0+0~mr6.4.0.0_all.deb
ngcp-rtpengine-daemon_6.4.0.0+0~mr6.4.0.0_amd64.deb
ngcp-rtpengine-iptables_6.4.0.0+0~mr6.4.0.0_amd64.deb
 ngcp-rtpengine-kernel-dkms_6.4.0.0+0~mr6.4.0.0_all.deb
ngcp-rtpengine-kernel-source_6.4.0.0+0~mr6.4.0.0_all.deb
ngcp-rtpengine-recording-daemon_6.4.0.0+0~mr6.4.0.0_amd64.deb
ngcp-rtpengine-utils_6.4.0.0+0~mr6.4.0.0_all.deb
Is it the reason of my issue?

On Fri, Jun 29, 2018 at 9:57 PM, Aqs Younas  wrote:
> you can check by starting your rtpengine with this parameter "--table=0". If
> kernal modules is properly installed you will not face any error.
>
> Best Regards,
>
> Aqs Younas
>
>
>
> On Fri, 29 Jun 2018 at 03:33, Mojtaba  wrote:
>>
>> Thank you. I installed RTPEngine from git
>> (https://github.com/sipwise/rtpengine).
>> Let me know how can i  be sure kernel module is loaded? and how can i load
>> it?
>> Thanks
>>
>> On Thu, Jun 28, 2018 at 6:20 PM, Richard Fuchs  wrote:
>> > Looks like you haven't set up the kernel module properly. For the "proc"
>> > method of recording to work, the kernel module must be loaded and in
>> > use.
>> >
>> > Cheers
>> >
>> >
>> > On 2018-06-28 05:47, Mojtaba wrote:
>> >>
>> >> Hi all,
>> >> I installed RTPEngine (Version: 6.4.0.0+0~mr6.4.0.0
>> >> git-master-4eb80da) in my VM machine (Debian GNU/Linux 8 (jessie), i
>> >> also installed kamailio 5.1.
>> >>
>> >> In kamailio routes, i have:
>> >> route {
>> >>  ...
>> >>  #rtpengine_manage("record-call")
>> >>  rtpengine_manage();
>> >>  start_recording();
>> >>  ...
>> >> }
>> >> Here is all other configuration files:
>> >>
>> >> In /etc/default/ngcp-rtpengine-daemon file:
>> >> RUN_RTPENGINE=yes
>> >> CONFIG_FILE=/etc/rtpengine/rtpengine.conf
>> >> CONFIG_SECTION=rtpengine
>> >> PIDFILE=/var/run/ngcp-rtpengine-daemon.pid
>> >> MANAGE_IPTABLES=yes
>> >> TABLE=0
>> >>
>> >> In /etc/default/ngcp-rtpengine-recording-daemon file:
>> >> RUN_RTPENGINE_RECORDING=yes
>> >> CONFIG_FILE=/etc/rtpengine/rtpengine-recording.conf
>> >> CONFIG_SECTION=rtpengine-recording
>> >> PIDFILE=/var/run/ngcp-rtpengine-recording-daemon.pid
>> >> MUST_NFS=no
>> >> NFS_HOST=192.168.1.1
>> >> NFS_REMOTE_PATH=/var/recordings
>> >> NFS_LOCAL_MOUNT=/var/lib/rtpengine-recording # must match output-dir if
>> >> used
>> >> NFS_OPTIONS=hard,intr,tcp
>> >>
>> >> In /etc/rtpengine/rtpengine.conf file:
>> >> [rtpengine]
>> >> table = 0
>> >> interface = 192.168.122.200
>> >> listen-ng = 127.0.0.1:2223
>> >> recording-dir = /var/spool/rtpengine/
>> >> recording-method = proc
>> >>
>> >> In
>> >> [rtpengine-recording]
>> >> table = 0
>> >> # output-storage = db (use default)
>> >> # output-format = mp3 (use default)
>> >> # output-mixed = 1
>> >> spool-dir = /var/spool/rtpengine/metadata
>> >> output-dir = /var/spool/rtpengine/recording
>> >>
>> >> But the recording is not work, I have these issues in syslog:
>> >> [1530178531.860171] INFO:
>> >> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command
>> >> 'start recording' from 127.0.0.1:38654
>> >> [1530178531.860191] NOTICE:
>> >> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Turning on call
>> >> recording.
>> >> [1530178531.860239] WARNING:
>> >> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Call recording through
>> >> /proc interface requested, but kernel table not open
>> >> [1530178531.860262] ERR:
>> >> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open
>> >> recording metadata file '(null)' for writing: Bad address
>> >> [1530178531.860269] ERR:
>> >> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open
>> >> recording metadata file '(null)' for writing: Bad address
>> >> [1530178531.860275] ERR:
>> >> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open
>> >> record

Re: [SR-Users] RTPEngine Recording does not work

2018-07-01 Thread Mojtaba
What do you mean?

On Sun, Jul 1, 2018 at 2:27 PM, Domenico Briganti  wrote:
> ll
>
> Il 01 lug 2018 11:48 AM, "Mojtaba"  ha scritto:
>
> I have strange issue,
> the output of of `uname -r` command is "3.16.0-4-amd64", while all the
> debs file are "4.9.0-0.bpo.6-amd64".
> ngcp-rtpengine_6.4.0.0+0~mr6.4.0.0_all.deb
> ngcp-rtpengine-daemon_6.4.0.0+0~mr6.4.0.0_amd64.deb
> ngcp-rtpengine-iptables_6.4.0.0+0~mr6.4.0.0_amd64.deb
>  ngcp-rtpengine-kernel-dkms_6.4.0.0+0~mr6.4.0.0_all.deb
> ngcp-rtpengine-kernel-source_6.4.0.0+0~mr6.4.0.0_all.deb
> ngcp-rtpengine-recording-daemon_6.4.0.0+0~mr6.4.0.0_amd64.deb
> ngcp-rtpengine-utils_6.4.0.0+0~mr6.4.0.0_all.deb
> Is it the reason of my issue?
>
> On Fri, Jun 29, 2018 at 9:57 PM, Aqs Younas  wrote:
>> you can check by starting your rtpengine with this parameter "--table=0".
>> If
>> kernal modules is properly installed you will not face any error.
>>
>> Best Regards,
>>
>> Aqs Younas
>>
>>
>>
>> On Fri, 29 Jun 2018 at 03:33, Mojtaba  wrote:
>>>
>>> Thank you. I installed RTPEngine from git
>>> (https://github.com/sipwise/rtpengine).
>>> Let me know how can i  be sure kernel module is loaded? and how can i
>>> load
>>> it?
>>> Thanks
>>>
>>> On Thu, Jun 28, 2018 at 6:20 PM, Richard Fuchs 
>>> wrote:
>>> > Looks like you haven't set up the kernel module properly. For the
>>> > "proc"
>>> > method of recording to work, the kernel module must be loaded and in
>>> > use.
>>> >
>>> > Cheers
>>> >
>>> >
>>> > On 2018-06-28 05:47, Mojtaba wrote:
>>> >>
>>> >> Hi all,
>>> >> I installed RTPEngine (Version: 6.4.0.0+0~mr6.4.0.0
>>> >> git-master-4eb80da) in my VM machine (Debian GNU/Linux 8 (jessie), i
>>> >> also installed kamailio 5.1.
>>> >>
>>> >> In kamailio routes, i have:
>>> >> route {
>>> >>  ...
>>> >>  #rtpengine_manage("record-call")
>>> >>  rtpengine_manage();
>>> >>  start_recording();
>>> >>  ...
>>> >> }
>>> >> Here is all other configuration files:
>>> >>
>>> >> In /etc/default/ngcp-rtpengine-daemon file:
>>> >> RUN_RTPENGINE=yes
>>> >> CONFIG_FILE=/etc/rtpengine/rtpengine.conf
>>> >> CONFIG_SECTION=rtpengine
>>> >> PIDFILE=/var/run/ngcp-rtpengine-daemon.pid
>>> >> MANAGE_IPTABLES=yes
>>> >> TABLE=0
>>> >>
>>> >> In /etc/default/ngcp-rtpengine-recording-daemon file:
>>> >> RUN_RTPENGINE_RECORDING=yes
>>> >> CONFIG_FILE=/etc/rtpengine/rtpengine-recording.conf
>>> >> CONFIG_SECTION=rtpengine-recording
>>> >> PIDFILE=/var/run/ngcp-rtpengine-recording-daemon.pid
>>> >> MUST_NFS=no
>>> >> NFS_HOST=192.168.1.1
>>> >> NFS_REMOTE_PATH=/var/recordings
>>> >> NFS_LOCAL_MOUNT=/var/lib/rtpengine-recording # must match output-dir
>>> >> if
>>> >> used
>>> >> NFS_OPTIONS=hard,intr,tcp
>>> >>
>>> >> In /etc/rtpengine/rtpengine.conf file:
>>> >> [rtpengine]
>>> >> table = 0
>>> >> interface = 192.168.122.200
>>> >> listen-ng = 127.0.0.1:2223
>>> >> recording-dir = /var/spool/rtpengine/
>>> >> recording-method = proc
>>> >>
>>> >> In
>>> >> [rtpengine-recording]
>>> >> table = 0
>>> >> # output-storage = db (use default)
>>> >> # output-format = mp3 (use default)
>>> >> # output-mixed = 1
>>> >> spool-dir = /var/spool/rtpengine/metadata
>>> >> output-dir = /var/spool/rtpengine/recording
>>> >>
>>> >> But the recording is not work, I have these issues in syslog:
>>> >> [1530178531.860171] INFO:
>>> >> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command
>>> >> 'start recording' from 127.0.0.1:38654
>>> >> [1530178531.860191] NOTICE:
>>> >> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Turning on call
>>> >> recording.
>>> >> [1530178531.860239] WARNING:
>>> >> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Call recording through

Re: [SR-Users] rtpengine installation dependencies

2018-06-26 Thread Mojtaba
Thank you, I do it successfully.
I use these commands to install all dependencies:
apt-get -t jessie-backports install debhelper libmysqlclient-dev
iptables-dev libavcodec-dev libavfilter-dev libcurl4-openssl-dev
libevent-dev libglib2.0-dev libhiredis-dev libjson-glib-dev
libpcap0.8-dev libssl-dev  libxmlrpc-core-c3-dev markdown

apt-get -t jessie-backports install dkms module-assistant
With Regards. Mojtaba

On Mon, Jun 25, 2018 at 7:01 PM, Aqs Younas  wrote:
> Hello, Mojtaba.
>
> I don't remember the exact setups and unfortunately don't have access to
> server. But let me write few things I did which may help you.
> Added below line in source.list.
> 1. deb http://ftp.debian.org/debian jessie-backports main
> 2. apt-get update
> 3. apt-get -t jessie-backports install debhelper
> 4. Installed latest version of ffmpeg from source
>
> I was not able to successfully create .deb files but i was able to compile
> it manually.
> These links might help you.
>
> https://voipmagazine.wordpress.com/2015/02/17/rtpengine-compilation-and-installation-in-fedora-redhat/
> http://hrhashmi.blogspot.com/2016/06/rtpengine-installation-on-amazon-ami.html
>
> Br, Aqs
>
>
>
>
> On Mon, 25 Jun 2018 at 14:19, Mojtaba  wrote:
>>
>> Hi,
>> Please share the source.list for getting the lasted version of debhelper.
>> Thanks
>>
>> On Sat, Apr 7, 2018 at 2:45 AM, Aqs Younas  wrote:
>> > Thanks for the answer.
>> >
>> > I had installed debhelper from Debian package but it had Version:
>> > 9.20150101+deb8u2.
>> >
>> > root@debian-769mb-miami-01:/usr/src/rtpengine# dpkg -s debhelper | grep
>> > '^Version:'
>> > Version: 9.20150101+deb8u2
>> >
>> > Whereas rtpengine demands debhelper (>= 10~)
>> >
>> > That is why i was trying to install it from debs.
>> >
>> > Anyway, I updated my source.list and was able to get the lasted version
>> > of
>> > debhelpler.
>> >
>> > Thanks for your help.
>> >
>> >
>> >
>> >
>> > On 6 April 2018 at 20:45, Ulrich Henning  wrote:
>> >>
>> >> Hi Aqs,
>> >>
>> >>
>> >>
>> >> just use the corresponding build flag ‘export
>> >> DEB_BUILD_PROFILES="pkg.ngcp-rtpengine.nobcg729"’ as stated in the
>> >> readme
>> >> (https://github.com/sipwise/rtpengine) and your issue should be fixed.
>> >>
>> >>
>> >>
>> >> BR,
>> >>
>> >> Henning
>> >>
>> >>
>> >>
>> >> Von: sr-users [mailto:sr-users-boun...@lists.kamailio.org] Im Auftrag
>> >> von
>> >> Aqs Younas
>> >> Gesendet: Freitag, 6. April 2018 17:31
>> >> An: Kamailio (SER) - Users Mailing List 
>> >> Betreff: [SR-Users] rtpengine installation dependencies
>> >>
>> >>
>> >>
>> >> Greetings list.
>> >>
>> >>
>> >>
>> >> This might not be related to Kamailio but I thought someone would be
>> >> able
>> >> to give a hand in resolving dependency during the installation of
>> >> rtpengine
>> >> on Debian 8 Jessie.
>> >>
>> >>
>> >>
>> >> It is complaining about below dependencies.
>> >>
>> >>
>> >>
>> >> root@debian-769mb-miami-01:/usr/src/rtpengine# dpkg-buildpackage
>> >>
>> >> dpkg-buildpackage: source package ngcp-rtpengine
>> >>
>> >> dpkg-buildpackage: source version 6.3.0.0+0~mr6.3.0.0
>> >>
>> >> dpkg-buildpackage: source distribution unstable
>> >>
>> >> dpkg-buildpackage: source changed by Sipwise Jenkins Builder
>> >> 
>> >>
>> >> dpkg-buildpackage: host architecture amd64
>> >>
>> >>  dpkg-source --before-build rtpengine
>> >>
>> >> dpkg-source: info: using options from rtpengine/debian/source/options:
>> >> --extend-diff-ignore=.gitreview
>> >>
>> >> dpkg-checkbuilddeps: Unmet build dependencies: debhelper (>= 10~)
>> >> libbcg729-dev
>> >>
>> >> dpkg-buildpackage: warning: build dependencies/conflicts unsatisfied;
>> >> aborting
>> >>
>> >> dpkg-buildpackage: warning: (Use -d flag to override.)
>> >>
>> >>
>> >>
>> >> I tried to install debhelper from deb but complained from dh-autor

Re: [SR-Users] rtpengine installation dependencies

2018-06-25 Thread Mojtaba
Hi,
Please share the source.list for getting the lasted version of debhelper.
Thanks

On Sat, Apr 7, 2018 at 2:45 AM, Aqs Younas  wrote:
> Thanks for the answer.
>
> I had installed debhelper from Debian package but it had Version:
> 9.20150101+deb8u2.
>
> root@debian-769mb-miami-01:/usr/src/rtpengine# dpkg -s debhelper | grep
> '^Version:'
> Version: 9.20150101+deb8u2
>
> Whereas rtpengine demands debhelper (>= 10~)
>
> That is why i was trying to install it from debs.
>
> Anyway, I updated my source.list and was able to get the lasted version of
> debhelpler.
>
> Thanks for your help.
>
>
>
>
> On 6 April 2018 at 20:45, Ulrich Henning  wrote:
>>
>> Hi Aqs,
>>
>>
>>
>> just use the corresponding build flag ‘export
>> DEB_BUILD_PROFILES="pkg.ngcp-rtpengine.nobcg729"’ as stated in the readme
>> (https://github.com/sipwise/rtpengine) and your issue should be fixed.
>>
>>
>>
>> BR,
>>
>> Henning
>>
>>
>>
>> Von: sr-users [mailto:sr-users-boun...@lists.kamailio.org] Im Auftrag von
>> Aqs Younas
>> Gesendet: Freitag, 6. April 2018 17:31
>> An: Kamailio (SER) - Users Mailing List 
>> Betreff: [SR-Users] rtpengine installation dependencies
>>
>>
>>
>> Greetings list.
>>
>>
>>
>> This might not be related to Kamailio but I thought someone would be able
>> to give a hand in resolving dependency during the installation of rtpengine
>> on Debian 8 Jessie.
>>
>>
>>
>> It is complaining about below dependencies.
>>
>>
>>
>> root@debian-769mb-miami-01:/usr/src/rtpengine# dpkg-buildpackage
>>
>> dpkg-buildpackage: source package ngcp-rtpengine
>>
>> dpkg-buildpackage: source version 6.3.0.0+0~mr6.3.0.0
>>
>> dpkg-buildpackage: source distribution unstable
>>
>> dpkg-buildpackage: source changed by Sipwise Jenkins Builder
>> 
>>
>> dpkg-buildpackage: host architecture amd64
>>
>>  dpkg-source --before-build rtpengine
>>
>> dpkg-source: info: using options from rtpengine/debian/source/options:
>> --extend-diff-ignore=.gitreview
>>
>> dpkg-checkbuilddeps: Unmet build dependencies: debhelper (>= 10~)
>> libbcg729-dev
>>
>> dpkg-buildpackage: warning: build dependencies/conflicts unsatisfied;
>> aborting
>>
>> dpkg-buildpackage: warning: (Use -d flag to override.)
>>
>>
>>
>> I tried to install debhelper from deb but complained from dh-autoreconf
>> and that itself complained abount debhelpler. So, no one is being installed.
>>
>> Also, If someone can help me how to resolve dependency for  libbcg729-dev
>> too.
>>
>> Any pointer or hint or link to some installation guide would be more than
>> welcome.
>>
>>
>>
>> Br, Aqs.
>>
>>
>> ___
>> Kamailio (SER) - Users Mailing List
>> sr-users@lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
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> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>



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[SR-Users] Error in parse json documents in kamailio with jansson module

2018-01-07 Thread Mojtaba
Hello,
I used ndbmongo module in kamailio,
I used mongodb_find command to get some data from mongo, like this:

if(mongodb_find("sermongo", "Asterisk", "user", "{ \"number\" :
\"$fU\" }", "res")) {
xlog("L_INFO", "Data from mongodb is \"
$mongodb(res=>value) \" \n");
$var(data) = $mongodb(res=>value);
}
The result in syslog is like this:
Jan  8 07:23:36 ubuntu14 /usr/local/sbin/kamailio[12127]: {1 2 INVITE
MDRlMjRlNGVjZGY1Y2JkYjA5OGM2ZTg1ZTFjZTk0Nzg.} INFO: 

Re: [SR-Users] Kamailio fail to start with db_mongodb and ndb_mongodb modules

2018-01-09 Thread Mojtaba
Hi, Please check the definition of DBURL in your code. Before i had
the same problem, I solved this issue like this:
# *** Value defines - IDs used later in config
#!ifdef WITH_MYSQL
# - database URL - used to connect to database server by modules such
#
#as: auth_db, acc, usrloc, a.s.o.
#!define DBURL "mysql://openser:openserrw@localhost/openser"
#!endif

#!ifdef WITH_MONGODB
#!define MONGODB "mongodb://localhost/Kamailio"
#!endif
.
.
.
# - usrloc params -
/* enable DB persistency for location entries */
#!ifdef WITH_USRLOCDB
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "use_domain", MULTIDOMAIN)
#!ifdef WITH_MONGODB
modparam("usrloc", "db_url", MONGODB)
#!else
modparam("usrloc", "db_url", DBURL)
#!endif
#!endif

With Regards.
Mojtaba Esfandiari.S

On Mon, Jan 8, 2018 at 12:36 PM, Abdul Basit <basit.e...@gmail.com> wrote:
> Hi Team,
>
> Any comments/ guidelines?
>
> --
> regards,
>
> abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445
>
> On 6 January 2018 at 00:39, Abdul Basit <basit.e...@gmail.com> wrote:
>>
>> Hi Daniel & K-Team,
>>
>> I setup kamailio 5.0.5 from git on vanilla Debian GNU/Linux 8.3 (jessie)
>>
>> # kamailio -v
>> version: kamailio 5.0.5 (x86_64/linux)
>> flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE,
>> USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC,
>> TLSF_MALLOC, DBG_SR_MEMORY, 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 8MB
>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>> id: unknown
>> compiled on 16:53:42 Jan  5 2018 with gcc 4.9.2
>>
>>
>> Installed mongodb from debian repo
>>
>> # dpkg -l | grep mongodb
>> ii  mongodb1:3.2.17-1amd64
>> object/document-oriented database (metapackage)
>> ii  mongodb-clients1:3.2.17-1amd64
>> object/document-oriented database (client apps)
>> ii  mongodb-dev1:2.4.10-5+deb8u1 amd64
>> object/document-oriented database (development)
>> ii  mongodb-server 1:3.2.17-1amd64
>> object/document-oriented database (server package)
>>
>> # mongo --version
>> MongoDB shell version: 3.2.17
>>
>> Will try sometime with mongodb latest version 3.6 as well as I will get
>> time.
>>
>> Installed libmongoc driver through sid repo "deb
>> http://ftp.de.debian.org/debian sid main"
>>
>> # dpkg -l | grep libmongoc
>> ii  libmongoc-1.0-0  1.9.0+dfsg-1
>> amd64MongoDB C client library - runtime files
>> ii  libmongoc-dev1.9.0+dfsg-1
>> amd64MongoDB C client library - dev files
>>
>> Kamailio compiled with
>>
>> # make cfg include_modules="utils app_lua json evapi db_mongodb
>> ndb_mongodb"
>>
>> Created version collection in mongodb
>>
>> # cat
>> /usr/local/src/kamailio/utils/kamctl/mongodb/kamailio/version-create.mongo |
>> mongo
>>
>> Setup kamailio.cfg as per https://telerain.com/kamailio
>>
>> This time kamailio started successfully without any crash :)
>>
>> -
>>
>> Now its time to use mongodb for kamailio backend database.
>>
>> Used default kamailio configuration and defined mongodb connection as
>>
>> #!define DBURL "mongodb://localhost/kamailio"
>>
>> I enabled WITH_USRLOCDB [#!define WITH_USRLOCDB], kamailio refuse to start
>> with errors.
>>
>> Jan  5 19:16:05 kmaongo kamailio: CRITICAL:  [core/cfg.y:3411]:
>> yyerror_at(): parse error in config file
>> /usr/local/etc/kamailio/kamailio.cfg, line 365, column 30-34: syntax error
>> Jan  5 19:16:05 kmaongo kamailio: CRITICAL:  [core/cfg.y:3411]:
>> yyerror_at(): parse error in config file
>> /usr/local/etc/kamailio/kamailio.cfg, line 365, column 30-34: Invalid
>> arguments
>> Jan  5 19:16:05 kmaongo kamailio: CRITICAL:  [core/cfg.y:3414]:
>> yyerror_at(): parse error in config file
>> /usr/local/etc/kamailio/kamailio.cfg, line 365, column 35:
>> Jan  5 19:16:05 kmaongo kamailio[28067]: ERROR: bad config file (3 errors)
>> Jan  5 19:16:05 kmaongo kamailio: WARNING:  [core/ppcfg.c:221]:
>> pp_ifdef_level_check(): different number of preprocessor directiv

Re: [SR-Users] SEMS with sbc application

2018-01-27 Thread Mojtaba
Hello,
Yes. The log is for sems with mo profile.
I  used SEMS as sbc applications in my network. Let me paste my
configurations below:
#sems.conf
  interfaces=intern,extern

  sip_ip_intern=192.168.18.20
  sip_port_intern=5060
  media_ip_intern=192.168.18.20
  rtp_low_port_intern=2000
  rtp_high_port_intern=5000

  sip_ip_extern=94.182.110.10
  sip_port_extern=4080
  media_ip_extern=94.182.110.10
  rtp_low_port_extern=2000
  rtp_high_port_extern=5000
  public_ip_extern=94.182.110.10
  #sig_sock_opts_extern=force_via_address
  tcp_connect_timeout_extern=1000
  tcp_idle_timeout_extern=90
I was forced to disable "sig_sock_opts_extern" option,

#sbc.conf is like this:

profiles=refuse_with_200,register,mo,mt,refuse
active_profile=$M($m=>methodmap),$M($si=>src_ipmap),refuse
regex_maps=src_ipmap,methodmap

#mo profile is like this:

# defaults: transparent
#RURI=$r
#From=$f
#To=$t
#Contact=<sip:$Ri>
#Call-ID
Call-ID=$ci_leg2
## routing
# outbound proxy:
#outbound_proxy=sip:192.168.5.106:5060
outbound_proxy=sip:192.168.18.19:5060

# force outbound proxy (in-dialog requests)?
#force_outbound_proxy=yes
force_outbound_proxy=yes

# destination IP[:port] for outgoing requests
#next_hop=192.168.5.106:5060
next_hop=192.168.18.19:5060

# set RURI to (calculated) next_hop
#patch_ruri_next_hop=yes
# update next_hop from remote destination? (e.g. from SRV)
#next_hop_fixed=yes
# outbound interface to use (interface ID)
outbound_interface=intern

# SIP NAT handling: recommended if dealing with far end NATs
dlg_nat_handling=yes

## RTP relay
# enable RTP relaying (bridging):
enable_rtprelay=yes
# force symmetric RTP (start with passive mode):
rtprelay_force_symmetric_rtp=yes

# RTP interface to use for A leg
aleg_rtprelay_interface=extern

# RTP interface to use for B leg
rtprelay_interface=intern

#mt profile is like this:

#RURI=$r
#From=$f
#To=$t

#Contact=<sip:$Ri>

#Call-ID
Call-ID=$ci_leg2

## routing
# outbound proxy:
#outbound_proxy=sip:192.168.5.106:5060
#outbound_proxy=sip:$H(P-Route)
#outbound_proxy=sip:$H(P-Source-IP):$H(P-Source-Port)

# force outbound proxy (in-dialog requests)?
force_outbound_proxy=yes
# destination IP[:port] for outgoing requests
#next_hop=192.168.5.106:5060
# set RURI to (calculated) next_hop
#patch_ruri_next_hop=yes
# update next_hop from remote destination? (e.g. from SRV)
#next_hop_fixed=yes
# outbound interface to use (interface ID)
outbound_interface=extern

# SIP NAT handling: recommended if dealing with far end NATs
dlg_nat_handling=yes

## RTP relay
# enable RTP relaying (bridging):
enable_rtprelay=yes
# force symmetric RTP (start with passive mode):
rtprelay_force_symmetric_rtp=yes

# RTP interface to use for A leg
aleg_rtprelay_interface=intern

# RTP interface to use for B leg
rtprelay_interface=extern

## filters:
## filters:
header_filter=blacklist
#header_list=P-App-Param,P-App-Name,P-Route,Remote-Party-ID
header_list=P-Route,Remote-Party-ID,P-Source-IP,P-Source-Port

I think it is big challenge if i am dependent to NAT between the uac
and SEMS (SIP ALG) when i changed default port on sbc.
How can i solved this problem in this regards?
Thanks. With Regards.Mojtaba



On Thu, Jan 25, 2018 at 7:31 PM, Stefan Sayer
<stefan.sa...@googlemail.com> wrote:
> Hello,
>
> Mojtaba wrote on 24.01.2018 12:17:
>> Hello,
>> I have a problem today, It's strange for me.
>> Suppose we have this senario:
>> uac1--->SEMS(mo profile)--->Kamailio>SEMS(mt
>> profile)-->uac2
>>
>> In above topology, we have two interfaces(intern,extern) for SEMS, and
>> just used as SBC (sbc application).
>> if i used port=5060 as external port, every things is right and log
>> file is like this:
>>  [#7fed3f9f9700/32820] [run, udp_trsp.cpp:352] DEBUG: vv M [|] u recvd
>> msg via UDP from 89.165.117.125:42411 vv
>> --++--
>> REGISTER sip:kava.shatel.ir;transport=UDP SIP/2.0
>> Via: SIP/2.0/UDP
>> 89.165.117.125:42411;branch=z9hG4bK-d8754z-021bd8b61efc7ac0-1---d8754z-
>> Max-Forwards: 70
>> Contact: <sip:4000@
>> 89.165.117.125:42411;rinstance=79011092e56e1a09;transport=UDP>
>> To: <sip:4...@kava.shatel.ir;transport=UDP>
>> From: <sip:4...@kava.shatel.ir;transport=UDP>;tag=82820e1f
>> Call-ID: Y2U4YThiYjEwNTUzMzliZTIwNWZkMDI3MTM4OTZlNWU.
>> CSeq: 2 REGISTER
>> Expires: 3600
>> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
>> INFO, SUBSCRIBE
>> Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
>> User-Agent: Z 3.3.25608 r25552
>> Allow-Events: presence, kpml
>> Content-Length: 0
>>
>> but when i  changed port=4080 for external port, The Via header and
>> contact header are changed to my public ip, like this:
>>
>>  [#7fed3f9f9700/32820] [run, udp_trsp.cpp:352] DEB

Re: [SR-Users] What is the mechanism used for sending push notification from Kamailio, For both IOS and Android?

2018-02-12 Thread Mojtaba
Hello Arunbalan,
Please read about TSILO module In kamailio, Also there is simple way
to implement concept of push notification as you needed.
With Regards.
Mojtaba Esfandiari.S

On Tue, Feb 13, 2018 at 10:19 AM, Arun NV <arunbala...@gmail.com> wrote:
> I am creating custom phone client, which uses Kamailio as a proxy and it's
> required to send push notification from Kamailio to the clients when a call
> towards the client reached on Kamailio. I found a module for APNS here
> https://github.com/tvntsr/push. But I am developing both Android and IOS
> clients. So What mechanism I can use for the Android side?
>
> Is there any module available for GCM(Google Cloud Messaging) in Kamailio?
>
> Thank you,
> Arunbalan
>
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Re: [SR-Users] NGN and NNI functionalities

2018-07-27 Thread Mojtaba
Helll,
The NGN  is the concept in next generation network.
The NNI (network to network interface) is deal with to conect two or more
networks with together.
The Kamailio has some modules to impelement IMS network, Also the IMS
 network could be connect with other IMS/Non IMS networks. For example some
nodes e.g. ICSCF, MGCF, BGCF could be used as NNI interface in IMS/NGN
networks.
With Regards.Mojtaba

On Fri, 27 Jul 2018 14:13 Duarte Rocha,  wrote:

> Greetings,
>
> Does Kamailio has any module that deals with NGN and NNI ?
>
> Also, is there any variable or transformation that gives me access to a
> parameter in the User Part of an URI ?
>
> Best Regards,
>
> Duarte Rocha
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Re: [SR-Users] Nat issue

2018-08-10 Thread Mojtaba
Hello eyas,
The NAT issue in IMS (and VoIP) solution networks, is affected to
user-plan (e.g Voice) only. In this regards, you maybe have no-way
audio or one-way audio.
As you mentions, Your isuue is in control-plan (signalling in IMS),
Then it is not regards of NAT issue.
Please check your network again. also check mediation devices between
p-cscf and user2's device, so that why the user2 doesn’t receive
INVITE signalling.
Do you could use Multi-Homed in p-cscf or not?
With Regards.Mojtaba

On Thu, Aug 9, 2018 at 1:11 AM, eyas barhouk  wrote:
> hello dears
>
> when i'm using kamailio 5.2 as IMS i'm facing the following NAT issue :
>
> when p-cscf handle a call from user1 to user2 that pre-registered on private
> address "192.X.X.X" (on another LAN) the user can't receive the invite
> message .
>
> in another word:
> proxy ip : 10.192.X.X
> user1 ip : 10.X.X.X
> user2 ip : 192.X.X.X
> the call scenario : user1 ---invite> user2 (not arrived)
>
> when p-cscf handle a call from user2 that pre-registered on private IP
> "192.X.X.X" (on another LAN) to  user1 the user receive the invite message
> .
>
> the call scenario : user2 ---invite> user1 (arrived)
>
> when both of the user on different LANs the message 200 OK doesn't arrive to
> proxy :\
>
> so based on that does any one has an experiment with NAT handling &
> nathelper module and how to configure it on P-cscf .
>
> thanks in advance
>
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Re: [SR-Users] Kamailio as a P-CSCF user

2018-08-16 Thread Mojtaba
If you want the kamailio works just as P-CSCF, you should add and
install these modules in kamailio:
ims_dialog module,
ims_usrloc_pcscf module,
ims_registrar_pcscf module

If you want use RX interface (between PCSCF and PCRF), These modules is needed:
cdp, cdp_avp, ims_qos

With Regards. Mojtaba

On Thu, Aug 16, 2018 at 10:35 AM, serkan celik  wrote:
> Hi Mojtaba,
> Thanks for your answer.
> Actually I know what you mean. In my question I skipped all the steps of
> signaling.
> Since my question is to be able to use Kamailio as a P-CSCF node.
>
> So I guess we can install I-CSCF, S-CFCF and HSS in seperate machines (VMs).
>
> I am already aware of the signaling of IMS entities. But I wonder how can I
> use kamailio as a P-CSCF node. and which modules should I install?
> I want to install kamailio as a PCSCF on ubuntu18.
> When I look at some IMS kamailio integration I see all IMS modules are
> installed. BTW I also tried to install IMS modules on my ubuntu18 but I got
> some errors during loading some modules as follow:
>
> ERROR:  [sr_module.c:582]: load_module(): could not open module
> :
> /usr/local/kamailio-4.4/lib64/kamailio/modules/cdp.so: undefined symbol:
> get_str_hash
>
>
> ws_frame.c:32:10: fatal error: unistr.h: No such file or directory
>  #include 
>   ^~
> compilation terminated.
> ../../Makefile.rules:97: recipe for target 'ws_frame.o' failed
>
>
> Thanks & Regards,
>
>
> On Thu, Aug 16, 2018 at 7:42 AM, Mojtaba  wrote:
>>
>> Hello Serkan,
>> In IMS concept, In while Registration flow, the scscf is selected by
>> I-CSCF node. Then, When the UE is registered in IMS (SCSCF node), The
>> SCSCF node would be appear as Service_Record header in SIP body.
>> The UE is aware about the it's SIP serving in IMS, and when it
>> originate new call, the PCSCF forward it's call to desired node.
>> For more information refer to 3gpp documents.
>> With Regards.Mojtaba
>>
>> On Wed, Aug 15, 2018 at 4:42 PM, serkan celik 
>> wrote:
>> > Hi all,
>> > I want to use Kamailio as a P-CSCF on ubuntu 18.
>> > Kamailio will be p-cscf and will communicate scscf in another open-ims
>> > box.
>> >
>> > How can I implement this feature. I have already looked following link.
>> > However it did not help me since I could not see any example pcscf
>> > configuration files after installation
>> >
>> > https://www.kamailio.org/wiki/tutorials/ims/installation-howto
>> >
>> > Regards,
>> > Serkan
>> >
>> > ___
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>> > sr-users@lists.kamailio.org
>> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>> >
>>
>>
>>
>> --
>> --Mojtaba Esfandiari.S
>>
>> ___
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>
>
>
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Re: [SR-Users] Kamailio as a P-CSCF user

2018-08-16 Thread Mojtaba
As the Henning was mentioned,
Try to use Kamailio V.5.X. i used Kamailio 5.0 before without any issue.
With Regards.Mojtaba

On Thu, Aug 16, 2018 at 11:50 AM, serkan celik  wrote:
> Yes I am sure that there is cdp.so lib in the module path
>
> On Thu, Aug 16, 2018 at 9:04 AM, Mojtaba  wrote:
>>
>> Do you sure the cdp module is installed successfully? Is there cdp.so
>> file in the module path in your system?
>>
>> On Thu, Aug 16, 2018 at 11:06 AM, serkan celik 
>> wrote:
>> > Hi Mojbata,
>> > I have installed all the modules you mentioned but when I start kamailio
>> > I
>> > am getting following error even I have cdp.so in modules path
>> >
>> >  ERROR:  [sr_module.c:582]: load_module(): could not open module
>> > :
>> > /usr/local/kamailio-4.4/lib64/kamailio/modules/cdp.so: undefined symbol:
>> > get_str_hash
>> >
>> >
>> > On Thu, Aug 16, 2018 at 8:22 AM, Mojtaba  wrote:
>> >>
>> >> If you want the kamailio works just as P-CSCF, you should add and
>> >> install these modules in kamailio:
>> >> ims_dialog module,
>> >> ims_usrloc_pcscf module,
>> >> ims_registrar_pcscf module
>> >>
>> >> If you want use RX interface (between PCSCF and PCRF), These modules is
>> >> needed:
>> >> cdp, cdp_avp, ims_qos
>> >>
>> >> With Regards. Mojtaba
>> >>
>> >> On Thu, Aug 16, 2018 at 10:35 AM, serkan celik 
>> >> wrote:
>> >> > Hi Mojtaba,
>> >> > Thanks for your answer.
>> >> > Actually I know what you mean. In my question I skipped all the steps
>> >> > of
>> >> > signaling.
>> >> > Since my question is to be able to use Kamailio as a P-CSCF node.
>> >> >
>> >> > So I guess we can install I-CSCF, S-CFCF and HSS in seperate machines
>> >> > (VMs).
>> >> >
>> >> > I am already aware of the signaling of IMS entities. But I wonder how
>> >> > can I
>> >> > use kamailio as a P-CSCF node. and which modules should I install?
>> >> > I want to install kamailio as a PCSCF on ubuntu18.
>> >> > When I look at some IMS kamailio integration I see all IMS modules
>> >> > are
>> >> > installed. BTW I also tried to install IMS modules on my ubuntu18 but
>> >> > I
>> >> > got
>> >> > some errors during loading some modules as follow:
>> >> >
>> >> > ERROR:  [sr_module.c:582]: load_module(): could not open module
>> >> > :
>> >> > /usr/local/kamailio-4.4/lib64/kamailio/modules/cdp.so: undefined
>> >> > symbol:
>> >> > get_str_hash
>> >> >
>> >> >
>> >> > ws_frame.c:32:10: fatal error: unistr.h: No such file or directory
>> >> >  #include 
>> >> >   ^~
>> >> > compilation terminated.
>> >> > ../../Makefile.rules:97: recipe for target 'ws_frame.o' failed
>> >> >
>> >> >
>> >> > Thanks & Regards,
>> >> >
>> >> >
>> >> > On Thu, Aug 16, 2018 at 7:42 AM, Mojtaba  wrote:
>> >> >>
>> >> >> Hello Serkan,
>> >> >> In IMS concept, In while Registration flow, the scscf is selected by
>> >> >> I-CSCF node. Then, When the UE is registered in IMS (SCSCF node),
>> >> >> The
>> >> >> SCSCF node would be appear as Service_Record header in SIP body.
>> >> >> The UE is aware about the it's SIP serving in IMS, and when it
>> >> >> originate new call, the PCSCF forward it's call to desired node.
>> >> >> For more information refer to 3gpp documents.
>> >> >> With Regards.Mojtaba
>> >> >>
>> >> >> On Wed, Aug 15, 2018 at 4:42 PM, serkan celik 
>> >> >> wrote:
>> >> >> > Hi all,
>> >> >> > I want to use Kamailio as a P-CSCF on ubuntu 18.
>> >> >> > Kamailio will be p-cscf and will communicate scscf in another
>> >> >> > open-ims
>> >> >> > box.
>> >> >> >
>> >> >> > How can I implement this feature. I have already looked following
>> >> >> > link.
>> >> >> > However it did not help me since I could not see any

Re: [SR-Users] Kamailio as a P-CSCF user

2018-08-15 Thread Mojtaba
Hello Serkan,
In IMS concept, In while Registration flow, the scscf is selected by
I-CSCF node. Then, When the UE is registered in IMS (SCSCF node), The
SCSCF node would be appear as Service_Record header in SIP body.
The UE is aware about the it's SIP serving in IMS, and when it
originate new call, the PCSCF forward it's call to desired node.
For more information refer to 3gpp documents.
With Regards.Mojtaba

On Wed, Aug 15, 2018 at 4:42 PM, serkan celik  wrote:
> Hi all,
> I want to use Kamailio as a P-CSCF on ubuntu 18.
> Kamailio will be p-cscf and will communicate scscf in another open-ims box.
>
> How can I implement this feature. I have already looked following link.
> However it did not help me since I could not see any example pcscf
> configuration files after installation
>
> https://www.kamailio.org/wiki/tutorials/ims/installation-howto
>
> Regards,
> Serkan
>
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Re: [SR-Users] Kamailio as a P-CSCF user

2018-08-16 Thread Mojtaba
Do you sure the cdp module is installed successfully? Is there cdp.so
file in the module path in your system?

On Thu, Aug 16, 2018 at 11:06 AM, serkan celik  wrote:
> Hi Mojbata,
> I have installed all the modules you mentioned but when I start kamailio I
> am getting following error even I have cdp.so in modules path
>
>  ERROR:  [sr_module.c:582]: load_module(): could not open module
> :
> /usr/local/kamailio-4.4/lib64/kamailio/modules/cdp.so: undefined symbol:
> get_str_hash
>
>
> On Thu, Aug 16, 2018 at 8:22 AM, Mojtaba  wrote:
>>
>> If you want the kamailio works just as P-CSCF, you should add and
>> install these modules in kamailio:
>> ims_dialog module,
>> ims_usrloc_pcscf module,
>> ims_registrar_pcscf module
>>
>> If you want use RX interface (between PCSCF and PCRF), These modules is
>> needed:
>> cdp, cdp_avp, ims_qos
>>
>> With Regards. Mojtaba
>>
>> On Thu, Aug 16, 2018 at 10:35 AM, serkan celik 
>> wrote:
>> > Hi Mojtaba,
>> > Thanks for your answer.
>> > Actually I know what you mean. In my question I skipped all the steps of
>> > signaling.
>> > Since my question is to be able to use Kamailio as a P-CSCF node.
>> >
>> > So I guess we can install I-CSCF, S-CFCF and HSS in seperate machines
>> > (VMs).
>> >
>> > I am already aware of the signaling of IMS entities. But I wonder how
>> > can I
>> > use kamailio as a P-CSCF node. and which modules should I install?
>> > I want to install kamailio as a PCSCF on ubuntu18.
>> > When I look at some IMS kamailio integration I see all IMS modules are
>> > installed. BTW I also tried to install IMS modules on my ubuntu18 but I
>> > got
>> > some errors during loading some modules as follow:
>> >
>> > ERROR:  [sr_module.c:582]: load_module(): could not open module
>> > :
>> > /usr/local/kamailio-4.4/lib64/kamailio/modules/cdp.so: undefined symbol:
>> > get_str_hash
>> >
>> >
>> > ws_frame.c:32:10: fatal error: unistr.h: No such file or directory
>> >  #include 
>> >   ^~
>> > compilation terminated.
>> > ../../Makefile.rules:97: recipe for target 'ws_frame.o' failed
>> >
>> >
>> > Thanks & Regards,
>> >
>> >
>> > On Thu, Aug 16, 2018 at 7:42 AM, Mojtaba  wrote:
>> >>
>> >> Hello Serkan,
>> >> In IMS concept, In while Registration flow, the scscf is selected by
>> >> I-CSCF node. Then, When the UE is registered in IMS (SCSCF node), The
>> >> SCSCF node would be appear as Service_Record header in SIP body.
>> >> The UE is aware about the it's SIP serving in IMS, and when it
>> >> originate new call, the PCSCF forward it's call to desired node.
>> >> For more information refer to 3gpp documents.
>> >> With Regards.Mojtaba
>> >>
>> >> On Wed, Aug 15, 2018 at 4:42 PM, serkan celik 
>> >> wrote:
>> >> > Hi all,
>> >> > I want to use Kamailio as a P-CSCF on ubuntu 18.
>> >> > Kamailio will be p-cscf and will communicate scscf in another
>> >> > open-ims
>> >> > box.
>> >> >
>> >> > How can I implement this feature. I have already looked following
>> >> > link.
>> >> > However it did not help me since I could not see any example pcscf
>> >> > configuration files after installation
>> >> >
>> >> > https://www.kamailio.org/wiki/tutorials/ims/installation-howto
>> >> >
>> >> > Regards,
>> >> > Serkan
>> >> >
>> >> > ___
>> >> > Kamailio (SER) - Users Mailing List
>> >> > sr-users@lists.kamailio.org
>> >> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> --Mojtaba Esfandiari.S
>> >>
>> >> ___
>> >> Kamailio (SER) - Users Mailing List
>> >> sr-users@lists.kamailio.org
>> >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>> >
>> >
>> >
>> > ___
>> > Kamailio (SER) - Users Mailing List
>> > sr-users@lists.kamailio.org
>> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>> >
>>
>>
>>
>> --
>> --Mojtaba Esfandiari.S
>>
>> ___
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>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
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[SR-Users] Issue with RTPEngine(6.5.0.0+0~mr6.5.0.0)

2018-08-16 Thread Mojtaba
Hello,
Before i used RTPEngine (6.4.0.0+0~mr6.4.0.0) without any problem,
Nowday i install RTPEngine from git. I notice that it's varsion is
changed to 6.5.0.0+0~mr6.5.0.0. When i want install
ngcp-rtpengine-recording-daemon, I have this issue:
Failed to start ngcp-rtpengine-recording-daemon.service: Unit
rpcbind.socket failed to load: No such file or directory.
invoke-rc.d: initscript ngcp-rtpengine-recording-daemon, action "start" failed.
dpkg: error processing package ngcp-rtpengine-recording-daemon (--install):
 subprocess installed post-installation script returned error exit status 6
Processing triggers for systemd (215-17+deb8u7) ...
Errors were encountered while processing:
 ngcp-rtpengine-recording-daemon

Please let me what is this issue?


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Re: [SR-Users] Issue with RTPEngine(6.5.0.0+0~mr6.5.0.0)

2018-08-16 Thread Mojtaba
Yes, the old version that work correct, it's version is "Version:
6.4.0.0+0~mr6.4.0.0 git-master-a9ec666"
But if i get from git, it's version is " 6.5.0.0+0~mr6.5.0.0." this
version does not install successfully.

On Thu, Aug 16, 2018 at 1:57 PM, Nicolas Breuer
 wrote:
> Hello,
>
> It's not an answer on your question but related to RPTEngine.
> Do you have any version number when you type rtpengine -v
>
> [root@proxy2 daemon]# ./rtpengine -v
> Version: undefined
>
> Is it expected that the version is undefined ?
>
> Thanks
>
>
> -Message d'origine-
> De : sr-users  De la part de Mojtaba
> Envoyé : jeudi 16 août 2018 11:23
> À : Kamailio (SER) - Users Mailing List 
> Objet : [SR-Users] Issue with RTPEngine(6.5.0.0+0~mr6.5.0.0)
>
> Hello,
> Before i used RTPEngine (6.4.0.0+0~mr6.4.0.0) without any problem, Nowday i 
> install RTPEngine from git. I notice that it's varsion is changed to 
> 6.5.0.0+0~mr6.5.0.0. When i want install ngcp-rtpengine-recording-daemon, I 
> have this issue:
> Failed to start ngcp-rtpengine-recording-daemon.service: Unit rpcbind.socket 
> failed to load: No such file or directory.
> invoke-rc.d: initscript ngcp-rtpengine-recording-daemon, action "start" 
> failed.
> dpkg: error processing package ngcp-rtpengine-recording-daemon (--install):
>  subprocess installed post-installation script returned error exit status 6 
> Processing triggers for systemd (215-17+deb8u7) ...
> Errors were encountered while processing:
>  ngcp-rtpengine-recording-daemon
>
> Please let me what is this issue?
>
>
> --
> --Mojtaba Esfandiari.S
>
> ___
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Re: [SR-Users] Registered User limit

2018-08-16 Thread Mojtaba
Hello,
When ever the UE is registered in Kamailio, One entry on Linux ARP
cache is fiiled. The default size of Linux ARP cache size in 3 levels
are:
gc_thresh1:
The minimum number of entries to keep in the ARP cache. The garbage
collector will not run if there are fewer than this number of entries
in the cache. Defaults to 128.
gc_thresh2:
The soft maximum number of entries to keep in the ARP cache. The
garbage collector will allow the number of entries to exceed this for
5 seconds before collection will be performed. Defaults to 512.
gc_thresh3:
The hard maximum number of entries to keep in the ARP cache. The
garbage collector will always run if there are more than this number
of entries in the cache. Defaults to 1024.

Then you should update the above parameters in your linux, like this:
echo 512 > /proc/sys/net/ipv4/neigh/default/gc_thresh1
echo 1024 > /proc/sys/net/ipv4/neigh/default/gc_thresh2
echo 2048 > /proc/sys/net/ipv4/neigh/default/gc_thresh3
These changes will remain until the next reboot. To make the changes
permanent, add the following lines to your /etc/sysctl.conf file.
net.ipv4.neigh.default.gc_thresh1 = 512
net.ipv4.neigh.default.gc_thresh2 = 1024
net.ipv4.neigh.default.gc_thresh3 = 2048

With Regards.Mojtabaa

On Thu, Aug 16, 2018 at 3:49 PM, Alex Balashov
 wrote:
> That sounds uncannily close to an open file descriptor limit.
>
> --
> Sent from mobile. Apologies for brevity and errors.
>
>
> -Original Message-
> From: Philip Fleck 
> To: Kamailio 
> Sent: Thu, 16 Aug 2018 7:11 AM
> Subject: [SR-Users] Registered User limit
>
> Is there a limit to the number of users which can be registered on Kamailio
> at any one time, or a configuration file which sets this.
>
> I have just installed Kamailio in a lab environment for testing purposes but
> I seem to only be able to register 1024 users, all remaining Register
> requests are ignored.
> I am not using a database to authenticate users or anything, just want to be
> able to register upto 10,000 users.
>
> regards
> Philip
>
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Re: [SR-Users] Issue with RTPEngine(6.5.0.0+0~mr6.5.0.0)

2018-08-16 Thread Mojtaba
Yes, it is installed, But doesn't work.

On Thu, Aug 16, 2018 at 4:28 PM, Richard Fuchs  wrote:
> On 2018-08-16 05:23, Mojtaba wrote:
>>
>> Hello,
>> Before i used RTPEngine (6.4.0.0+0~mr6.4.0.0) without any problem,
>> Nowday i install RTPEngine from git. I notice that it's varsion is
>> changed to 6.5.0.0+0~mr6.5.0.0. When i want install
>> ngcp-rtpengine-recording-daemon, I have this issue:
>> Failed to start ngcp-rtpengine-recording-daemon.service: Unit
>> rpcbind.socket failed to load: No such file or directory.
>> invoke-rc.d: initscript ngcp-rtpengine-recording-daemon, action "start"
>> failed.
>> dpkg: error processing package ngcp-rtpengine-recording-daemon
>> (--install):
>>   subprocess installed post-installation script returned error exit status
>> 6
>> Processing triggers for systemd (215-17+deb8u7) ...
>> Errors were encountered while processing:
>>   ngcp-rtpengine-recording-daemon
>>
>> Please let me what is this issue?
>>
>>
>
> Do you have the rpcbind package installed?
>
>
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[SR-Users] remove_hf function in INVITE message

2018-08-19 Thread Mojtaba
Today i encounter with strange issue in SIP signalling in Kamailio,
In INVITE message, I just change Contact Header and relay message to
destination, But the message is like this:
192.168.122.245.4080 > 192.168.122.174.5060: [bad udp cksum 0x7b5e ->
0x3a4d!] SIP, length: 1104
INVITE sip:8...@r-kh.com;transport=UDP SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP
192.168.122.245:4080;branch=z9hG4bK14d7.dc974309aba269b24213204d3ea085e2.0
Via: SIP/2.0/UDP
192.168.122.1:36853;rport=36853;branch=z9hG4bK-d8754z-423b431a3ebc8a22-1---d8754z-
Max-Forwards: 69
sip:7000@192.168.122.1:36853;transport=UDP;alias=192.168.122.1~36853~1To:

From: ;tag=8e47e95e
Call-ID: YTZmYTJmYzg0ZDNiMjM2OGUxZGYwNzg4OWY1ZmIxYzc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Allow-Events: presence, kpml
Content-Length: 241
 
The code in Kamailio is:

#!ifdef WITH_SBC_NASIM
#if the Invite request is MO, forward it to Asterisk .
if (is_method("INVITE") && $si!="192.168.122.245") {
xlog("L_INFO", "The Invite message from $si:$sp,
Forwart to Asterisk.  <192.168.122.174:5060>.\n");
remove_hf("Contact");
append_hf("Contact:sip:$f...@sbc.r-kh.com\r\n");
$du = "sip:192.168.122.174:5060";
}
#!endif

Be notice that, i have the same procedure in REGISTER message, But it
is work correct,
if(is_method("REGISTER")) {
xlog("L_INFO", "forward register to Asterisk\n");
remove_hf("Contact");
append_hf("Contact:sip:$f...@sbc.r-kh.com\r\n");
rewriteuri("sip:r-kh.com");
$du = "sip:192.168.122.174:5060";
route(RELAY);
}

Let me know what is the problem?

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[SR-Users] Using SEMS in IMS-ZTE Network

2018-08-26 Thread Mojtaba
EFER, MESSAGE, OPTIONS,
INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Allow-Events: presence, kpml
Contact: 
;expires=3600
Content-Length: 0


08:23:32.790414 IP (tos 0x0, ttl 255, id 35790, offset 0, flags
[none], proto UDP (17), length 593)
172.23.1.1.5080 > 172.23.1.252.5080: [udp sum ok] UDP, length 565
E..Q.=..SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.23.1.252:5080;rport=5080;branch=z9hG4bK~0uk0ahN
To: 
;tag=ztesipDEmH2AWT*3-5-16648*egei.3
From: 
;tag=34384541-5B814A44000AE296-4B5F5700
Call-ID: 37B52BC0-5B814A44000AE592-4B5F5700
CSeq: 10 REGISTER
Content-Length: 0
WWW-Authenticate: Digest
realm="r-kh.com",domain="sip:scscf1.r-kh.com",nonce="179797f920a09d65b5641e5e87df0d8c",opaque="aW1zLmNvbS5jbg==",stale=TRUE,algorithm=MD5,qop="auth"


08:23:32.855067 IP (tos 0x0, ttl 64, id 17364, offset 0, flags [DF],
proto UDP (17), length 1071)
172.23.1.252.5080 > 172.23.1.1.5080: [bad udp cksum 0x5f58 ->
0xf553!] UDP, length 1043
E../C.@.@._X

REGISTER sip:r-kh.com SIP/2.0
Via: SIP/2.0/UDP 172.23.1.252:5080;branch=z9hG4bKeAfKkaZa;rport
From: 
;tag=5A64C27D-5B814A44000CD034-4B6F6700
To: 
CSeq: 10 REGISTER
Call-ID: 5380C97E-5B814A44000CD2A5-4B6F6700
Route: 
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest
username="+985137054...@r-kh.com",realm="r-kh.com",nonce="179797f920a09d65b5641e5e87df0d8c",uri="sip:r-kh.com;transport=UDP",response="017efb986e17506a82bb01e8eb136b90",cnonce="3764fd5eb2ea9539d61ec14196a2bf00",nc=0001,qop=auth,algorithm=MD5,opaque="aW1zLmNvbS5jbg=="
Allow-Events: presence, kpml
Contact: 
;expires=3600
Content-Length: 0


08:23:32.860417 IP (tos 0x0, ttl 255, id 35804, offset 0, flags
[none], proto UDP (17), length 508)
172.23.1.1.5080 > 172.23.1.252.5080: [udp sum ok] UDP, length 480
E...SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.23.1.252:5080;rport=5080;branch=z9hG4bKeAfKkaZa
To: 
;tag=ztesipxxsD3FR2VG8e9zxR_1dx*1-1-16648*ihad.1
From: 
;tag=5A64C27D-5B814A44000CD034-4B6F6700
Call-ID: 5380C97E-5B814A44000CD2A5-4B6F6700
CSeq: 10 REGISTER
X-ZTE-Cause: "CSCF-1.1.BC004016-BC004288-0-0.pcscf1.r-kh.com"
Retry-After: 30
Content-Length: 0




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Re: [SR-Users] remove_hf function in INVITE message

2018-08-19 Thread Mojtaba
Hi again,
I used remove_hf_re and textop module ( subst('/re/repl/flags')) too,
like bellow:
   if(is_method("INVITE")){
   
subst('/^Cantact:(.*)$/Contact:sip:$f...@sbc.r-khorasan.tci.ir/ig');
   #or
   remove_hf("Contact");
   #or
   append_hf("Contact:sip:$f...@sbc.r-khorasan.tci.ir\r\n");
   }
But does not work, The "Bad Sip Request" is received.
On Sun, Aug 19, 2018 at 2:15 PM Mojtaba  wrote:
>
> Today i encounter with strange issue in SIP signalling in Kamailio,
> In INVITE message, I just change Contact Header and relay message to
> destination, But the message is like this:
> 192.168.122.245.4080 > 192.168.122.174.5060: [bad udp cksum 0x7b5e ->
> 0x3a4d!] SIP, length: 1104
> INVITE sip:8...@r-kh.com;transport=UDP SIP/2.0
> Record-Route: 
> Via: SIP/2.0/UDP
> 192.168.122.245:4080;branch=z9hG4bK14d7.dc974309aba269b24213204d3ea085e2.0
> Via: SIP/2.0/UDP
> 192.168.122.1:36853;rport=36853;branch=z9hG4bK-d8754z-423b431a3ebc8a22-1---d8754z-
> Max-Forwards: 69
> sip:7000@192.168.122.1:36853;transport=UDP;alias=192.168.122.1~36853~1To:
> 
> From: ;tag=8e47e95e
> Call-ID: YTZmYTJmYzg0ZDNiMjM2OGUxZGYwNzg4OWY1ZmIxYzc.
> CSeq: 1 INVITE
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
> INFO, SUBSCRIBE
> Content-Type: application/sdp
> Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
> User-Agent: Z 3.3.25608 r25552
> Allow-Events: presence, kpml
> Content-Length: 241
>  
> The code in Kamailio is:
>
> #!ifdef WITH_SBC_NASIM
> #if the Invite request is MO, forward it to Asterisk .
> if (is_method("INVITE") && $si!="192.168.122.245") {
> xlog("L_INFO", "The Invite message from $si:$sp,
> Forwart to Asterisk.  <192.168.122.174:5060>.\n");
> remove_hf("Contact");
> append_hf("Contact:sip:$f...@sbc.r-kh.com\r\n");
> $du = "sip:192.168.122.174:5060";
> }
> #!endif
>
> Be notice that, i have the same procedure in REGISTER message, But it
> is work correct,
> if(is_method("REGISTER")) {
> xlog("L_INFO", "forward register to Asterisk\n");
>     remove_hf("Contact");
> append_hf("Contact:sip:$f...@sbc.r-kh.com\r\n");
> rewriteuri("sip:r-kh.com");
> $du = "sip:192.168.122.174:5060";
> route(RELAY);
> }
>
> Let me know what is the problem?
>
> --
> --Mojtaba Esfandiari.S



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Re: [SR-Users] help

2018-08-18 Thread Mojtaba
Hi,
The installation of Kamailio is straight forward, Just follow the
following steps in this site:
https://www.kamailio.org/wiki/install/stable/debian
Also, you could download Kamailio form git source.
With Regards.Mojtaba

On Sat, Aug 18, 2018 at 11:52 AM Pravin .  wrote:
>
> Hello Team,
>
> I want to install Kamailio SIP server on centOS server...do we need to 
> download kamailio ISO file and install on server..pls provide the 
> installation procedure/guidelines.
>
>
> Regards
> Pravin
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Re: [SR-Users] help

2018-08-18 Thread Mojtaba
Hi,
1-It is regard of your OS, I think the Debian is better.
2-It doesn't need download iso, You could install it from git source.
3- The kamailio has a lot of modules, you could add your favourite
modules from source.lst file. It is very straight forward.
With Regards.Mojtaba
On Sat, Aug 18, 2018 at 12:51 PM Pravin .  wrote:
>
> Thanks  Mojtaba.
>
> I'm installing Kamailio 1st time so i have approach you for help in this 
> regard.
>
> Just want to know what are the pre-requisite for installing kamailio like 
>
> 1. What is the operating system require? ( i have centOS 7 installed in my 
> server)
> 2. do we need to download kamailio ISO file and burn it to installed in centOS
> 3. I checked on site and in download option , i have downloaded 
> "kamailio-5.1.4_src" , how will i installed with this file ?
>
> Need your guidance/support  for installation and commissioning of Kamailio 
> SIP server.
>
> Regards,
> Pravin
> bolindia Networks Pvt. Ltd
>
>
>
>
>
>
>
> On Sat, Aug 18, 2018 at 1:38 PM, Mojtaba  wrote:
>>
>> Hi,
>> The installation of Kamailio is straight forward, Just follow the
>> following steps in this site:
>> https://www.kamailio.org/wiki/install/stable/debian
>> Also, you could download Kamailio form git source.
>> With Regards.Mojtaba
>>
>> On Sat, Aug 18, 2018 at 11:52 AM Pravin .  wrote:
>> >
>> > Hello Team,
>> >
>> > I want to install Kamailio SIP server on centOS server...do we need to 
>> > download kamailio ISO file and install on server..pls provide the 
>> > installation procedure/guidelines.
>> >
>> >
>> > Regards
>> > Pravin
>> > ___
>> > Kamailio (SER) - Users Mailing List
>> > sr-users@lists.kamailio.org
>> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>
>> --
>> --Mojtaba Esfandiari.S
>>
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>
>
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Re: [SR-Users] TOPOH module im REGISTER method.

2018-09-05 Thread Mojtaba
Hi all,
I try to used SEMS in this regards, But i have this issue.
https://lists.kamailio.org/pipermail/sr-users/2018-August/102753.html
Any updates?
Thanks
On Tue, Sep 4, 2018 at 2:07 PM Mojtaba  wrote:
>
> Hello,
> I am used topoh module in kamailio, Why this module does not work (No
> change ) in Register request Method?
> I need to change UA' ip address in Via header in Register request
> before forwarded to my network? Does the topoh module used in this
> situation?
> I have not this issue in INVITE request, I just have this issue in
> Register request.
> Thanks.
> With Regards.Mojtaba
>
> --
> --Mojtaba Esfandiari.S



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Re: [SR-Users] tpops module

2018-09-10 Thread Mojtaba
Hi,
In regard of w_tcp_keepalive_disable1() function in tcpops module,
this function called a macro to convert string to integer.
Let show the value of $avp(bye_conid) after set. Be sure the it's
value is string id.
Thanks with regards.
Mojtaba
On Tue, Sep 11, 2018 at 3:21 AM  wrote:
>
> Hello Henning,
> I can reproduce this issues on each BYE.
> Not sure if this related.
>
>
> Sep 10 17:51:40 canlvprx01 /usr/sbin/kamailio[20421]: CRITICAL: 
> [core/route.c:1202]: comp_num(): Invalid right operand (1)
>
>
> volga629
>
> On Mon, Sep 10, 2018 at 3:56 AM, Henning Westerholt 
> wrote:
> > Am Samstag, 8. September 2018, 17:28:33 CEST schrieb
> > volga...@networklab.ca:
> >>  I am using tcpops module  for keepalive and on BYE when tcp
> >> keeplaive
> >>  close is called produce this error in log.
> >>
> >>  Sep  8 11:21:26 canlvprx01 /usr/sbin/kamailio[18933]: ERROR: 
> >>  [core/sr_module.c:1781]: get_int_fparam(): Could not convert PV to
> >> int
> >>  Sep  8 11:21:26 canlvprx01 /usr/sbin/kamailio[18933]: ERROR: tcpops
> >>  [tcpops_mod.c:293]: w_tcp_keepalive_disable1(): invalid parameter
> >> 'con'
> >>  (must be a number)
> >>
> >>  kamailio-5.1.3-1.gitf0dce0c99.fc27.x86_64
> >>
> >>
> >>  if (is_method("BYE|CANCEL")) {
> >>  $avp(bye_conid) = $conid;
> >>  rtpengine_delete();
> >>  t_on_reply("4");
> >>   }
> >>
> >>  onreply_route[4] {
> >>  if(is_method("BYE") && t_check_status("200")) {
> >>  tcp_keepalive_disable("$avp(bye_conid)");
> >>  }
> >>  }
> >
> > Hello,
> >
> > this error is produced because the $conid PV could not converted to an
> > integer. Do you see some other error before this one? Is this error
> > happens
> > all the time or just for rare or special cases?
> >
> > Best regards,
> >
> > Henning
> >
> >
> > --
> > Henning Westerholt
> > https://skalatan.de/blog/
> >
>
>
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Re: [SR-Users] kamailio + rtpengine + custom filename

2018-09-10 Thread Mojtaba
Hello,
You could use metadata to save your favourite filename on it. After
recording is finished, replace it. The metadata file is created based
on  for each call.
With Regards.
Mojtaba
On Mon, Sep 10, 2018 at 8:53 PM Жан Базаров  wrote:
>
> Hello! Filenames have the format --.
>  How i can change filename in start_recording() ???
>
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Re: [SR-Users] rtpengine installation dependencies

2018-07-05 Thread Mojtaba
In regard of my last post, If you have issue about kernel module, run
this command to update it:
apt-get install -t jessie-backports linux-image-amd64
Thanks

On Tue, Jun 26, 2018 at 1:43 PM, Mojtaba  wrote:
> Thank you, I do it successfully.
> I use these commands to install all dependencies:
> apt-get -t jessie-backports install debhelper libmysqlclient-dev
> iptables-dev libavcodec-dev libavfilter-dev libcurl4-openssl-dev
> libevent-dev libglib2.0-dev libhiredis-dev libjson-glib-dev
> libpcap0.8-dev libssl-dev  libxmlrpc-core-c3-dev markdown
>
> apt-get -t jessie-backports install dkms module-assistant
> With Regards. Mojtaba
>
> On Mon, Jun 25, 2018 at 7:01 PM, Aqs Younas  wrote:
>> Hello, Mojtaba.
>>
>> I don't remember the exact setups and unfortunately don't have access to
>> server. But let me write few things I did which may help you.
>> Added below line in source.list.
>> 1. deb http://ftp.debian.org/debian jessie-backports main
>> 2. apt-get update
>> 3. apt-get -t jessie-backports install debhelper
>> 4. Installed latest version of ffmpeg from source
>>
>> I was not able to successfully create .deb files but i was able to compile
>> it manually.
>> These links might help you.
>>
>> https://voipmagazine.wordpress.com/2015/02/17/rtpengine-compilation-and-installation-in-fedora-redhat/
>> http://hrhashmi.blogspot.com/2016/06/rtpengine-installation-on-amazon-ami.html
>>
>> Br, Aqs
>>
>>
>>
>>
>> On Mon, 25 Jun 2018 at 14:19, Mojtaba  wrote:
>>>
>>> Hi,
>>> Please share the source.list for getting the lasted version of debhelper.
>>> Thanks
>>>
>>> On Sat, Apr 7, 2018 at 2:45 AM, Aqs Younas  wrote:
>>> > Thanks for the answer.
>>> >
>>> > I had installed debhelper from Debian package but it had Version:
>>> > 9.20150101+deb8u2.
>>> >
>>> > root@debian-769mb-miami-01:/usr/src/rtpengine# dpkg -s debhelper | grep
>>> > '^Version:'
>>> > Version: 9.20150101+deb8u2
>>> >
>>> > Whereas rtpengine demands debhelper (>= 10~)
>>> >
>>> > That is why i was trying to install it from debs.
>>> >
>>> > Anyway, I updated my source.list and was able to get the lasted version
>>> > of
>>> > debhelpler.
>>> >
>>> > Thanks for your help.
>>> >
>>> >
>>> >
>>> >
>>> > On 6 April 2018 at 20:45, Ulrich Henning  wrote:
>>> >>
>>> >> Hi Aqs,
>>> >>
>>> >>
>>> >>
>>> >> just use the corresponding build flag ‘export
>>> >> DEB_BUILD_PROFILES="pkg.ngcp-rtpengine.nobcg729"’ as stated in the
>>> >> readme
>>> >> (https://github.com/sipwise/rtpengine) and your issue should be fixed.
>>> >>
>>> >>
>>> >>
>>> >> BR,
>>> >>
>>> >> Henning
>>> >>
>>> >>
>>> >>
>>> >> Von: sr-users [mailto:sr-users-boun...@lists.kamailio.org] Im Auftrag
>>> >> von
>>> >> Aqs Younas
>>> >> Gesendet: Freitag, 6. April 2018 17:31
>>> >> An: Kamailio (SER) - Users Mailing List 
>>> >> Betreff: [SR-Users] rtpengine installation dependencies
>>> >>
>>> >>
>>> >>
>>> >> Greetings list.
>>> >>
>>> >>
>>> >>
>>> >> This might not be related to Kamailio but I thought someone would be
>>> >> able
>>> >> to give a hand in resolving dependency during the installation of
>>> >> rtpengine
>>> >> on Debian 8 Jessie.
>>> >>
>>> >>
>>> >>
>>> >> It is complaining about below dependencies.
>>> >>
>>> >>
>>> >>
>>> >> root@debian-769mb-miami-01:/usr/src/rtpengine# dpkg-buildpackage
>>> >>
>>> >> dpkg-buildpackage: source package ngcp-rtpengine
>>> >>
>>> >> dpkg-buildpackage: source version 6.3.0.0+0~mr6.3.0.0
>>> >>
>>> >> dpkg-buildpackage: source distribution unstable
>>> >>
>>> >> dpkg-buildpackage: source changed by Sipwise Jenkins Builder
>>> >> 
>>> >>
>>> >> dpkg-buildpackage: host architecture amd64
>>> >>
>>>

Re: [SR-Users] RTPEngine Recording does not work

2018-07-05 Thread Mojtaba
I solved it, it just update kernel module.
apt-get install -t jessie-backports linux-image-amd64
Thanks

On Sun, Jul 1, 2018 at 2:37 PM, Mojtaba  wrote:
> What do you mean?
>
> On Sun, Jul 1, 2018 at 2:27 PM, Domenico Briganti  wrote:
>> ll
>>
>> Il 01 lug 2018 11:48 AM, "Mojtaba"  ha scritto:
>>
>> I have strange issue,
>> the output of of `uname -r` command is "3.16.0-4-amd64", while all the
>> debs file are "4.9.0-0.bpo.6-amd64".
>> ngcp-rtpengine_6.4.0.0+0~mr6.4.0.0_all.deb
>> ngcp-rtpengine-daemon_6.4.0.0+0~mr6.4.0.0_amd64.deb
>> ngcp-rtpengine-iptables_6.4.0.0+0~mr6.4.0.0_amd64.deb
>>  ngcp-rtpengine-kernel-dkms_6.4.0.0+0~mr6.4.0.0_all.deb
>> ngcp-rtpengine-kernel-source_6.4.0.0+0~mr6.4.0.0_all.deb
>> ngcp-rtpengine-recording-daemon_6.4.0.0+0~mr6.4.0.0_amd64.deb
>> ngcp-rtpengine-utils_6.4.0.0+0~mr6.4.0.0_all.deb
>> Is it the reason of my issue?
>>
>> On Fri, Jun 29, 2018 at 9:57 PM, Aqs Younas  wrote:
>>> you can check by starting your rtpengine with this parameter "--table=0".
>>> If
>>> kernal modules is properly installed you will not face any error.
>>>
>>> Best Regards,
>>>
>>> Aqs Younas
>>>
>>>
>>>
>>> On Fri, 29 Jun 2018 at 03:33, Mojtaba  wrote:
>>>>
>>>> Thank you. I installed RTPEngine from git
>>>> (https://github.com/sipwise/rtpengine).
>>>> Let me know how can i  be sure kernel module is loaded? and how can i
>>>> load
>>>> it?
>>>> Thanks
>>>>
>>>> On Thu, Jun 28, 2018 at 6:20 PM, Richard Fuchs 
>>>> wrote:
>>>> > Looks like you haven't set up the kernel module properly. For the
>>>> > "proc"
>>>> > method of recording to work, the kernel module must be loaded and in
>>>> > use.
>>>> >
>>>> > Cheers
>>>> >
>>>> >
>>>> > On 2018-06-28 05:47, Mojtaba wrote:
>>>> >>
>>>> >> Hi all,
>>>> >> I installed RTPEngine (Version: 6.4.0.0+0~mr6.4.0.0
>>>> >> git-master-4eb80da) in my VM machine (Debian GNU/Linux 8 (jessie), i
>>>> >> also installed kamailio 5.1.
>>>> >>
>>>> >> In kamailio routes, i have:
>>>> >> route {
>>>> >>  ...
>>>> >>  #rtpengine_manage("record-call")
>>>> >>  rtpengine_manage();
>>>> >>  start_recording();
>>>> >>  ...
>>>> >> }
>>>> >> Here is all other configuration files:
>>>> >>
>>>> >> In /etc/default/ngcp-rtpengine-daemon file:
>>>> >> RUN_RTPENGINE=yes
>>>> >> CONFIG_FILE=/etc/rtpengine/rtpengine.conf
>>>> >> CONFIG_SECTION=rtpengine
>>>> >> PIDFILE=/var/run/ngcp-rtpengine-daemon.pid
>>>> >> MANAGE_IPTABLES=yes
>>>> >> TABLE=0
>>>> >>
>>>> >> In /etc/default/ngcp-rtpengine-recording-daemon file:
>>>> >> RUN_RTPENGINE_RECORDING=yes
>>>> >> CONFIG_FILE=/etc/rtpengine/rtpengine-recording.conf
>>>> >> CONFIG_SECTION=rtpengine-recording
>>>> >> PIDFILE=/var/run/ngcp-rtpengine-recording-daemon.pid
>>>> >> MUST_NFS=no
>>>> >> NFS_HOST=192.168.1.1
>>>> >> NFS_REMOTE_PATH=/var/recordings
>>>> >> NFS_LOCAL_MOUNT=/var/lib/rtpengine-recording # must match output-dir
>>>> >> if
>>>> >> used
>>>> >> NFS_OPTIONS=hard,intr,tcp
>>>> >>
>>>> >> In /etc/rtpengine/rtpengine.conf file:
>>>> >> [rtpengine]
>>>> >> table = 0
>>>> >> interface = 192.168.122.200
>>>> >> listen-ng = 127.0.0.1:2223
>>>> >> recording-dir = /var/spool/rtpengine/
>>>> >> recording-method = proc
>>>> >>
>>>> >> In
>>>> >> [rtpengine-recording]
>>>> >> table = 0
>>>> >> # output-storage = db (use default)
>>>> >> # output-format = mp3 (use default)
>>>> >> # output-mixed = 1
>>>> >> spool-dir = /var/spool/rtpengine/metadata
>>>> >> output-dir = /var/spool/rtpengine/recording
>>>> >>
>>>> >&g

Re: [SR-Users] dispatcher module complete tutorial

2018-07-12 Thread Mojtaba
Hi,
You could use this tutorial - Load Balancing in OpenSIPS. It is
similar to Kamailio.
http://www.opensips.org/Documentation/Tutorials-LoadBalancing-1-9
With Regards
Mojtaba

On Wed, Jul 11, 2018 at 6:34 PM, SamSung TV  wrote:
> I want to use kamailio as load balancing server
>
>
>
> my topo:
>
> (Private IP) Kamailio  server2
>
> /
>
> /
> (public iP)Kamailio server1(load balance)(Private IP)
>
> \
> \
>
> (Private IP) Kamailio server1
>
> Any body can recomment me some tutorial to do this?
> Thanks any advance!
>
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Re: [SR-Users] RTPEngine-Recording-daemon

2018-07-09 Thread Mojtaba
I underestand your meaning.
Thank you.

On Mon, Jul 9, 2018 at 1:54 PM, Pawel Kuzak  wrote:
> Hi,
>
> You can. For every new call a new connection is established using the UNIX
> socket. So different connections means different calls.
>
> Regards,
> Paul
>
>
>
> Am 09.07.2018 um 10:43 schrieb Mojtaba:
>>
>> Hello,
>> In RTPEngine recording-daemon, I enabled "forward-to" option to
>> unix_socket. All things work right, I got metadata at first, then
>> media packets (RTP-UDP-IP).
>> The metadata for each calls is received just at first packet, but i
>> want to know, if i have more than one concurrent active call in
>> RTPEngine, How can i distinct which media packet was from which call?
>> Thanks with Regards.
>> --Mojtaba Esfandiari.S
>>
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>
>
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[SR-Users] RTPEngine-Recording-daemon

2018-07-09 Thread Mojtaba
Hello,
In RTPEngine recording-daemon, I enabled "forward-to" option to
unix_socket. All things work right, I got metadata at first, then
media packets (RTP-UDP-IP).
The metadata for each calls is received just at first packet, but i
want to know, if i have more than one concurrent active call in
RTPEngine, How can i distinct which media packet was from which call?
Thanks with Regards.
--Mojtaba Esfandiari.S

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[SR-Users] ASN.1 HI2- IRI-Parameters

2018-04-24 Thread Mojtaba
Hello.
I need to use HI2Operations for transfering IRI-Parameters in kamailio.Dose
anybody have experience in this regards?
Thanks.
Mojtaba
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Re: [SR-Users] Kamailio iOS Push Notifications

2018-03-03 Thread Mojtaba
Hello.
Please check tislo module in kamailio.
With regards.Mojtaba

On 1 Mar 2018 14:50, "Amine Mensori"  wrote:

> Hello,
>
> We need help implementing push notifications within Kamailio. I have found
> a module relating to this: https://github.com/tvntsr/push
> However, the developer did not provide any examples of the cfg file, only
> parts of certain functions. We need some help implementing the functions in
> our config file or a full example of another method to get push
> notifications to work when the application is closed on the device. Please
> note we are not very proficient in C programming so a clear example would
> help a lot.
>
> Many Thanks,
> Amine
>
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Re: [SR-Users] RTPEngine Recording does not work

2018-06-28 Thread Mojtaba
Thank you. I installed RTPEngine from git
(https://github.com/sipwise/rtpengine).
Let me know how can i  be sure kernel module is loaded? and how can i load it?
Thanks

On Thu, Jun 28, 2018 at 6:20 PM, Richard Fuchs  wrote:
> Looks like you haven't set up the kernel module properly. For the "proc"
> method of recording to work, the kernel module must be loaded and in use.
>
> Cheers
>
>
> On 2018-06-28 05:47, Mojtaba wrote:
>>
>> Hi all,
>> I installed RTPEngine (Version: 6.4.0.0+0~mr6.4.0.0
>> git-master-4eb80da) in my VM machine (Debian GNU/Linux 8 (jessie), i
>> also installed kamailio 5.1.
>>
>> In kamailio routes, i have:
>> route {
>>  ...
>>  #rtpengine_manage("record-call")
>>  rtpengine_manage();
>>  start_recording();
>>  ...
>> }
>> Here is all other configuration files:
>>
>> In /etc/default/ngcp-rtpengine-daemon file:
>> RUN_RTPENGINE=yes
>> CONFIG_FILE=/etc/rtpengine/rtpengine.conf
>> CONFIG_SECTION=rtpengine
>> PIDFILE=/var/run/ngcp-rtpengine-daemon.pid
>> MANAGE_IPTABLES=yes
>> TABLE=0
>>
>> In /etc/default/ngcp-rtpengine-recording-daemon file:
>> RUN_RTPENGINE_RECORDING=yes
>> CONFIG_FILE=/etc/rtpengine/rtpengine-recording.conf
>> CONFIG_SECTION=rtpengine-recording
>> PIDFILE=/var/run/ngcp-rtpengine-recording-daemon.pid
>> MUST_NFS=no
>> NFS_HOST=192.168.1.1
>> NFS_REMOTE_PATH=/var/recordings
>> NFS_LOCAL_MOUNT=/var/lib/rtpengine-recording # must match output-dir if
>> used
>> NFS_OPTIONS=hard,intr,tcp
>>
>> In /etc/rtpengine/rtpengine.conf file:
>> [rtpengine]
>> table = 0
>> interface = 192.168.122.200
>> listen-ng = 127.0.0.1:2223
>> recording-dir = /var/spool/rtpengine/
>> recording-method = proc
>>
>> In
>> [rtpengine-recording]
>> table = 0
>> # output-storage = db (use default)
>> # output-format = mp3 (use default)
>> # output-mixed = 1
>> spool-dir = /var/spool/rtpengine/metadata
>> output-dir = /var/spool/rtpengine/recording
>>
>> But the recording is not work, I have these issues in syslog:
>> [1530178531.860171] INFO:
>> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command
>> 'start recording' from 127.0.0.1:38654
>> [1530178531.860191] NOTICE:
>> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Turning on call
>> recording.
>> [1530178531.860239] WARNING:
>> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Call recording through
>> /proc interface requested, but kernel table not open
>> [1530178531.860262] ERR:
>> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open
>> recording metadata file '(null)' for writing: Bad address
>> [1530178531.860269] ERR:
>> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open
>> recording metadata file '(null)' for writing: Bad address
>> [1530178531.860275] ERR:
>> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open
>> recording metadata file '(null)' for writing: Bad address
>> [1530178531.860281] ERR:
>> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open
>> recording metadata file '(null)' for writing: Bad address
>> [1530178531.860287] ERR:
>> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open
>> recording metadata file '(null)' for writing: Bad address
>> [1530178531.860292] ERR:
>> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open
>> recording metadata file '(null)' for writing: Bad address
>> [1530178531.860303] INFO:
>> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Replying to 'start
>> recording' from 127.0.0.1:38654 (elapsed time 0.000112 sec)
>> [1530178531.969040] INFO:
>> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command
>> 'start recording' from 127.0.0.1:39374
>> [1530178531.969069] INFO:
>> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Replying to 'start
>> recording' from 127.0.0.1:39374 (elapsed time 0.04 sec)
>> [1530178537.176906] INFO:
>> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command
>> 'delete' from 127.0.0.1:40258
>> [1530178537.177003] INFO:
>> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Scheduling deletion of
>> call branch '9052b05a' (via-branch '') in 30 seconds
>> [1530178537.177026] INFO:
>> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Replying to 'delete'
>> from 127.0.0.1:40258 (elapsed time 0.84 sec)
>> [1530178537.178368] INFO:
>> [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command
>> 

[SR-Users] RTPEngine Recording does not work

2018-06-28 Thread Mojtaba
Hi all,
I installed RTPEngine (Version: 6.4.0.0+0~mr6.4.0.0
git-master-4eb80da) in my VM machine (Debian GNU/Linux 8 (jessie), i
also installed kamailio 5.1.

In kamailio routes, i have:
route {
...
#rtpengine_manage("record-call")
rtpengine_manage();
start_recording();
...
}
Here is all other configuration files:

In /etc/default/ngcp-rtpengine-daemon file:
RUN_RTPENGINE=yes
CONFIG_FILE=/etc/rtpengine/rtpengine.conf
CONFIG_SECTION=rtpengine
PIDFILE=/var/run/ngcp-rtpengine-daemon.pid
MANAGE_IPTABLES=yes
TABLE=0

In /etc/default/ngcp-rtpengine-recording-daemon file:
RUN_RTPENGINE_RECORDING=yes
CONFIG_FILE=/etc/rtpengine/rtpengine-recording.conf
CONFIG_SECTION=rtpengine-recording
PIDFILE=/var/run/ngcp-rtpengine-recording-daemon.pid
MUST_NFS=no
NFS_HOST=192.168.1.1
NFS_REMOTE_PATH=/var/recordings
NFS_LOCAL_MOUNT=/var/lib/rtpengine-recording # must match output-dir if used
NFS_OPTIONS=hard,intr,tcp

In /etc/rtpengine/rtpengine.conf file:
[rtpengine]
table = 0
interface = 192.168.122.200
listen-ng = 127.0.0.1:2223
recording-dir = /var/spool/rtpengine/
recording-method = proc

In
[rtpengine-recording]
table = 0
# output-storage = db (use default)
# output-format = mp3 (use default)
# output-mixed = 1
spool-dir = /var/spool/rtpengine/metadata
output-dir = /var/spool/rtpengine/recording

But the recording is not work, I have these issues in syslog:
[1530178531.860171] INFO:
[MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command
'start recording' from 127.0.0.1:38654
[1530178531.860191] NOTICE:
[MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Turning on call
recording.
[1530178531.860239] WARNING:
[MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Call recording through
/proc interface requested, but kernel table not open
[1530178531.860262] ERR:
[MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open
recording metadata file '(null)' for writing: Bad address
[1530178531.860269] ERR:
[MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open
recording metadata file '(null)' for writing: Bad address
[1530178531.860275] ERR:
[MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open
recording metadata file '(null)' for writing: Bad address
[1530178531.860281] ERR:
[MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open
recording metadata file '(null)' for writing: Bad address
[1530178531.860287] ERR:
[MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open
recording metadata file '(null)' for writing: Bad address
[1530178531.860292] ERR:
[MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open
recording metadata file '(null)' for writing: Bad address
[1530178531.860303] INFO:
[MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Replying to 'start
recording' from 127.0.0.1:38654 (elapsed time 0.000112 sec)
[1530178531.969040] INFO:
[MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command
'start recording' from 127.0.0.1:39374
[1530178531.969069] INFO:
[MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Replying to 'start
recording' from 127.0.0.1:39374 (elapsed time 0.04 sec)
[1530178537.176906] INFO:
[MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command
'delete' from 127.0.0.1:40258
[1530178537.177003] INFO:
[MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Scheduling deletion of
call branch '9052b05a' (via-branch '') in 30 seconds
[1530178537.177026] INFO:
[MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Replying to 'delete'
from 127.0.0.1:40258 (elapsed time 0.84 sec)
[1530178537.178368] INFO:
[MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command
'start recording' from 127.0.0.1:40258
[1530178537.178392] INFO:
[MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Replying to 'start
recording' from 127.0.0.1:40258 (elapsed time 0.04 sec)
^C[1530178543.752740] INFO: Version 6.4.0.0+0~mr6.4.0.0
git-master-4eb80da shutting down

Also when i start RTPEngine, I have  FATAL error about module
xt_RTPENGINEer in syslog:
Jun 28 05:44:58 debian8 systemd[1]: Starting NGCP RTP/media Proxy Daemon...
Jun 28 05:44:58 debian8 ngcp-rtpengine-iptables-setup[10203]:
modprobe: FATAL: Module xt_RTPENGINE not found.
Jun 28 05:44:58 debian8 ngcp-rtpengine-iptables-setup[10203]:
iptables: No chain/target/match by that name.
Jun 28 05:44:58 debian8 ngcp-rtpengine-iptables-setup[10203]:
ip6tables: No chain/target/match by that name.
Jun 28 05:44:58 debian8 systemd[1]: Started NGCP RTP/media Proxy Daemon.
Jun 28 05:44:58 debian8 rtpengine[10219]: [1530179098.989322] INFO:
Generating new DTLS certificate
Jun 28 05:44:59 debian8 rtpengine[10219]: [1530179099.033499] ERR:
FAILED TO CREATE KERNEL TABLE 0 (No such file or directory), KERNEL
FORWARDING DISABLED
J

Let me know what is the problem and how can i solve it?

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Re: [SR-Users] Crash ims_diameter_server

2018-10-31 Thread Mojtaba
Hi.
There are some reason for this issue.
The three nested json is not reason for crash ims_diameter.
Let try this senario.
Chenge type of the last indoor level of json from List to other type(like
int or ...).
Then check the lamailio is crashed or not?
With Regards.Mojtaba


On Wed, 31 Oct 2018 21:56 Iman Mohammadi, 
wrote:

> When the below format is sent from rest , kamailio crashes
>
> List[{
> List[{
> List[{
>  ]}
>  ]}
>  ]}
> (Json with 3 nested lists),
> With 2 lists it works properly,
> When diameter is translated to json with 3 lists by this module it also
> works properly,
> Json format is correct too.
>
> gdb:
>
> gdb) bt full
> #0  0x7f5e23c4a11e in diameterserver_add_avp (m=0x0, d=0x7f5e1b5c2240 "", 
> len=12, avp_code=431, flags=64, vendorid=0, data_do=2,
> func=0x7f5e23c51d6c <__FUNCTION__.20148> "parselist") at avp_helper.c:208
> avp = 0x7f5e1b5c3bd0
> __FUNCTION__ = "diameterserver_add_avp"
> #1  0x7f5e23c4d0c8 in parselist (response=0x0, list=0x7fff9f21a8d0, 
> item=0x11b67d0, level=2) at avp_helper.c:309
> i = 1
> flags = 64
> x = "p\250!\237"
> avp_list = {head = 0x0, tail = 0x0}
> avp_list_s = {s = 0x7f5e1b5c2240 "", len = 12}
> __FUNCTION__ = "parselist"
> #2  0x7f5e23c4cffc in parselist (response=0x7f5e1b5c3ab8, list=0x0, 
> item=0x11b6550, level=1) at avp_helper.c:304
> subitem = 0x11b67d0
> i = 0
> flags = 64
> x = "\000\000\000"
> avp_list = {head = 0x0, tail = 0x0}
> avp_list_s = {s = 0x7fff9f21a8f0 "@\251!\237\377\177", len = 
> 600070897}
> __FUNCTION__ = "parselist"
> #3  0x7f5e23c4db3a in addAVPsfromJSON (response=0x7f5e1b5c3ab8, 
> json=0x7f5e23e53950 ) at avp_helper.c:349
> subitem = 0x11b6550
> i = 4
> __FUNCTION__ = "addAVPsfromJSON"
> root = 0x11b4210
> #4  0x7f5e23c3fbdb in callback_cdp_request (request_in=0x7f5e1b5c19b0, 
> param=0x0) at ims_diameter_server.c:193
> fmsg = 0xab7840 <_faked_msg>
> backup_rt = 1
> ctx = {rec_lev = 0, run_flags = 0, last_retcode = 1, jmp_env = 
> {{__jmpbuf = {1, -5713032302318786866, 7971288, 7971288,
> 140042162528420, 0, -5713032302339758386, 
> 5712822384154044110}, __mask_was_saved = 0, __saved_mask = {__val = {
>   140735863172072, 127, 0, 140042162532340, 140042389578761, 
> 4222451713, 140042162532340, 140042162532340, 140042162532340,
>   140042162532340, 140042162532358, 140042162532467, 
> 140042162532340, 140042162532467, 0, 0}
> response = 0x7f5e1b5c3ab8
> __FUNCTION__ = "callback_cdp_request"
> #5  0x7f5e24a705c0 in api_callback (p=0x7f5e1b598f50, msg=0x7f5e1b5c19b0, 
> ptr=0x0) at api_process.c:83
> t = 0x7f5e1b598f50
> auto_drop = 32767
> h = 0x7f5e1b5b3358
> ---Type  to continue, or q  to quit---
> x = {type = (unknown: 2669784000), handler = {requestHandler = 
> 0x7f5e23c3eed1 ,
> responseHandler = 0x7f5e23c3eed1 }, param = 
> 0x3ee9f21ac10, next = 0x7f5e1b5988b8, prev = 0x19f210069}
> type = REQUEST_HANDLER
> rsp = 0x7f5e1b5c19b0
> __FUNCTION__ = "api_callback"
> #6  0x7f5e24a857d7 in worker_process (id=5) at worker.c:346
> t = {p = 0x7f5e1b598f50, msg = 0x7f5e1b5c19b0}
> cb = 0x7f5e1b59df30
> r = 128
> __FUNCTION__ = "worker_process"
> #7  0x7f5e24a62a8e in diameter_peer_start (blocking=0) at 
> diameter_peer.c:242
> pid = 0
> k = 5
> p = 0x36
> __FUNCTION__ = "diameter_peer_start"
> #8  0x7f5e24a559bc in cdp_child_init (rank=0) at cdp_mod.c:243
> __FUNCTION__ = "cdp_child_init"
> #9  0x00547e54 in init_mod_child (m=0x7f5e28656fb8, rank=0) at 
> core/sr_module.c:943
> __FUNCTION__ = "init_mod_child"
> #10 0x00547b16 in init_mod_child (m=0x7f5e28657c38, rank=0) at 
> core/sr_module.c:939
> __FUNCTION__ = "init_mod_child"
> #11 0x00547b16 in init_mod_child (m=0x7f5e28658010, rank=0) at 
> core/sr_module.c:939
> __FUNCTION__ = "init_mod_child"
> #12 0x00547b16 in init_mod_child (m=0x7f5e28658790, rank=0) at 
> core/sr_module.c:939
> __FUNCTION__ = "init_mod_child"
> #13 0x00547b16 in init_mod_child (m=0x7f5e28658d68, rank=0) at 
> core/sr_module.c:939
> __FUNCTION__ = "init_mod_child"
> #14 0x005481e6 in init_child (rank=0) at core/sr_module.c:970
> No locals.
> #15 0x004250f5 in main_loop () at main.c:1701
> i = 8
> pid = 1964
> si = 0x0
> si_desc = "udp receiver child=7 
> sock=10.25.5.23:5060\000y\000\000\000\000\000x\333i(^\177\000\000@\264!\237\377\177",
>  '\000' , 
> "`\263!\237\377\177\000\000\347L\340(^\177\000\000\360\260!\237\377\177\000\000\060\000\000\000\060\000\000\000\260\261!\237\377\177\000\000\350\226i(^\177\000"
> 

Re: [SR-Users] Installing rtpengine from RPM on CentOS

2018-11-02 Thread Mojtaba
Hello,
The installation of rtpengine is straightforward, But the installation
of dependencies may be different.
Please check this post:
https://lists.kamailio.org/pipermail/sr-users/2018-July/102114.html
I have write this for Debian.
With regards. Mojtaba
On Fri, Nov 2, 2018 at 12:27 AM Henning Westerholt  wrote:
>
> Am Donnerstag, 1. November 2018, 15:00:18 CET schrieb Pan Christensen:
> > That’s the kamailio module to control rtpengine, not the rtpengine itself.
> > [..]
>
> Hello Pan,
>
> in case you did not get an answer on this list I would suggest to contact the
> rtpengine authors directly.
>
> The page https://github.com/sipwise/rtpengine/blob/master/el/README.el.md
> indeed does not contain any informations about a Cent.OS repository.
>
> There was a discussion a few month ago (about manual installation on Cent.OS):
> https://lists.kamailio.org/pipermail/sr-users/2018-June/101952.html
>
> Installing from source or building your own RPMs is described as well on the
> docs quoted above, as you probably already read.
>
> Best regards,
>
> Henning
>
> --
> Henning Westerholt - https://skalatan.de/blog/
> Kamailio services - https://skalatan.de/services
> Kamailio security assessment - https://skalatan.de/de/assessment
>
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Re: [SR-Users] Parse Header error when send Register to P-CSCF using ims-pcscf

2018-11-02 Thread Mojtaba
Hello,
Would you paste the register flow SIP message in wireshark or tcpdump?
And Please paste the XML senario that you used to generate SIP
message.
Thank With Regards.Mojtaba
On Sun, Oct 28, 2018 at 10:20 PM Hưng Nguyễn Tiến
 wrote:
>
> Hi, i'm newbie in kamailio,
> I tried installed ims kamailio from ng-repository packages,
>
> Then when I tried to send register to SIP server using SIPp,
> I received this log
>  [Proxy-CSCF][28505]: ERROR:  [core/parser/msg_parser.c:331]: 
> parse_headers(): bad header field [(null)]
> [Proxy-CSCF][28500]: ERROR:  [core/parser/msg_parser.c:331]: 
> parse_headers(): bad header field [(null)]
>  [Proxy-CSCF][28496]: WARNING:  [core/receive.c:192]: receive_msg(): 
> parsing relevant headers failed
>  [Proxy-CSCF][28505]: message repeated 2 times: [ WARNING:  
> [core/receive.c:192]: receive_msg(): parsing relevant headers failed]
>  [Proxy-CSCF][28492]: ERROR:  [core/parser/msg_parser.c:675]: 
> parse_msg(): ERROR: parse_msg: 
> message=<#002#020#002#021#023�#023���z'��z'#006��[%�#007>
>
> Thank you
> --
> Nguyễn Tiến Hưng
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Re: [SR-Users] high availability and load balancing

2018-11-09 Thread Mojtaba
Hi,
The concept of HA in IMS is a little different in non-IMS network.
In each node of CSCF in IMS, you could do HA separately.
For example in PCSCF, you could use DNS load balancer to select one of
your PCSCF.  Actually the UE just see one domain as PCSCF, while you
may have more than one PCSCF in your network.
In each PCSCF, you could use dispatcher module to load balance all
Registration traffics between ICSCF nodes.  All initiate traffices
(like Invite message and etc) route to specific SCSCF via
service-route header.
In Registration message in ICSCF nodes, you could use HA to load
balance traffics between SCSCF in LIR Diameter message. For example in
LIA Diameter, you could sent a name of SCSCF thet has lower traffics.
finally the HA in IMS is doning some procedure, not only one or two task.
With Regards. Mojtaba Esfandiari.S
On Fri, Nov 9, 2018 at 11:23 PM Kjeld Flarup
 wrote:
>
> Hi Eyas
>
>
> A typical HA setup has two kamailio proxies with identical setup.
>
> Use keepalived or similar to give the proxies a shared IP.
>
> Because of the shared IP, all traffic will go to one proxy only.
>
> Thus each of the proxies must be able to handle all traffic, mening the 
> functionality must be very simple - no media handling.
>
> Then use the dispatcher module to do load balancing.
>
>
>
>
>  Med Liberalistiske Hilsner --
>Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog
>Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49
>Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk
>
> On 11/9/18 5:09 PM, eyas barhouk wrote:
>
> hello dears,
>
> i'm using kamailio 5.1 as IMS , and i have the following servers:
>
> p-cscf
> i-cscf-1 ,icscf-2
> s-cscf-1,scscf-2
>
> i need to do high availability and load balancing between my servers , but i 
> have some questions like do i have to use keepalived or dispatcher module and 
> what the benefits of each one of them ,and on load balancing method should i 
> do it for the register requests or only for invite requests or just for every 
> thing ??
> if any one has an idea or any thing to guide me i will be thankful
>
> thanks in advance and regards
>
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Re: [SR-Users] Core dump on diameter CCR request

2018-11-12 Thread Mojtaba
Hello,
We developed OCS based on these module in Kamailio: ims_charging,
ims_ocs, ims_dimameter_server and worker process written in Q_HSS. We
have not this issue.
Are you sure ims_charging is connected to any ocs proccess like ims_ocs?
Because CCR/CCA and other related message is transfered between them.
With Regards.
Mojtaba Esfandiari.S

On Mon, Nov 12, 2018 at 11:48 AM Daniel-Constantin Mierla
 wrote:
>
> Hello,
>
> get the full backtrace with gdb from the core file -- it should reveal
> where the crash happened.
>
> Also, it might be good to open an issue in bug tracker from
> github.com/kamailio/kamailio in order to track it properly.
>
> Cheers,
> Daniel
>
>
> On 11.11.18 12:52, Venkatesh Macha wrote:
> > Hi all,
> >
> > I am trying to implement Diameter charging with Kamailio. I am following
> > below tutorial
> > https://www.kamailio.org/events/2016-KamailioWorld/Day2/15-Carsten.Bock-Diameter-Charging-With-Kamailio.pdf
> >
> > But whenever I try to do the Ro_CCR from the INVITE block, Kamailio is
> > crashing and generating the core dump. I am new to diameter. I might be
> > doing something wrong here.
> >
> > please someone help me here.
> >
> > Please find the *diameter.xml* and *Diameter_core_dump_full_debug.txt* file.
> >
> > diameter.xml
> > <http://sip-router.1086192.n5.nabble.com/file/t6374/diameter.xml>
> > Diameter_core_dump_full_Debug.txt
> > <http://sip-router.1086192.n5.nabble.com/file/t6374/Diameter_core_dump_full_Debug.txt>
> >
> >
> > Here is my kamailio.cfg :
> >
> > loadmodule "cdp"
> > loadmodule "cdp_avp"
> > loadmodule "ims_charging"
> > loadmodule "ims_dialog"
> >
> > modparam("cdp","config_file","/etc/kamailio/diameter.xml")
> > modparam("cdp", "debug_heavy", 1)
> > # - ims_diameter_ro params -
> > #modparam("ims_charging", "single_ro_session_per_dialog", 1)
> > modparam("ims_charging", "origin_host", "10.195.0.38");
> > modparam("ims_charging", "origin_realm", "10.195.0.38");
> > modparam("ims_charging", "destination_realm", "mydomain.com");
> > modparam("ims_charging","interim_update_credits",30);
> > modparam("ims_charging","timer_buffer",5);
> > # -- ims_dialog params --
> > modparam("ims_dialog", "dlg_flag", 8)
> >
> >
> > While starting the kamailio, i am getting following errors:
> > Nov 11 11:40:03 ip-10-195-0-38 /usr/local/sbin/kamailio[25653]: ERROR: cdp
> > [peerstatemachine.c:675]: add_peer_application(): Too many applications for
> > this peer (max 0), not adding Application 4:0.
> >
> > Before crashing kamailio showing following error:
> >
> > Nov 11 11:39:49 ip-10-195-0-38 /usr/local/sbin/kamailio[25571]: INFO: {1
> > 18325 INVITE Nf6-dl2WeXF.LJhXotiRsuMhijsNx45o} 

Re: [SR-Users] registrar timer and database modes

2018-10-11 Thread Mojtaba
Hi,
In Kamailio version 5.1 or higher, you could see this configure
parameter like this:
modparam("ims_usrloc_scscf", "db_mode", 1), which used database to
save location on subscriber table. Maybe you should update your
kamailio.

When the subscriber deregister itself, it's record in database is
deleted logically, and needed some time to deleted from database.
With Regards.Mojtaba Esfandiari.S





On Wed, Oct 10, 2018 at 9:42 PM eyas barhouk  wrote:
>
> hello dears ,
>
> i'm using kamailio 5.0.1 as IMS platform and have the following issues :
>
> for Module "ims_usrloc_scscf" there is no documentation and i can't find a 
> way to set database mode =2 and define the tables .
> for Module "ims_registrar_scscf"when the user Unregistered it takes about 1~2 
> minutes to remove it from the memory on database mode 0 .
>
> so is there a way to minimize the timer when the user Unregistered and remove 
> it immediately from the memory  .
>
> thanks in advance & best regards.
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Re: [SR-Users] dispatcher probe

2018-09-04 Thread Mojtaba
Hi,
Sure, You could add in REQINIT route block in Kamailio.
With Regads.Mojtaba
On Tue, Sep 4, 2018 at 5:11 AM  wrote:
>
> Hello Everyone,
> Is possible access OPTIONS packet generated by dispatcher probe ?
> I am trying insert something like this.
>
> if(is_method("OPTIONS") && $hdr(User-Agent)=~"kamailio") {
> append_hf("X-Node-ID: $avp(node_id)\r\n");
> }
>
>
> volga629
>
>
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[SR-Users] TOPOH module im REGISTER method.

2018-09-04 Thread Mojtaba
Hello,
I am used topoh module in kamailio, Why this module does not work (No
change ) in Register request Method?
I need to change UA' ip address in Via header in Register request
before forwarded to my network? Does the topoh module used in this
situation?
I have not this issue in INVITE request, I just have this issue in
Register request.
Thanks.
With Regards.Mojtaba

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Re: [SR-Users] escape character

2018-09-24 Thread Mojtaba
Hello.
Try with this format:
$tU =~ "[/\/"/'/&]"
with regards.Mojtaba


On Tue, 25 Sep 2018 03:25 David Villasmil, 
wrote:

> Hello guys,
>
> I'd like to search for " ' or &
> But i can't seem to figure out how... I tried:
>
> $tU =~ "[\"'&]"
>
> But that's not catching it...
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
> ᐧ
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Re: [SR-Users] kamailio 5.0.6 | segfault bad msg or missing FROM header

2018-09-26 Thread Mojtaba
Hi,
As i see the dump.txt file, Please paste the SIP signalling to check
more. As the log, the From header may be malformed syntax.
because when parsing the From header, the issue is occurred. This
error is raised  when Form header isn't parse properly.
Also check the shared and private memory. May be in while parsing the
sip message, "out of memory"  is occurred to.
Whit Regards.
Mojtaba Esfandiari.S
On Wed, Sep 26, 2018 at 9:58 AM Laurent Schweizer
 wrote:
>
> Dear all,
>
>
>
> I have setup few weeks ago a kamailio 5.0.6, was working correctly till last 
> night where 2 time he stop with a segfault error :
>
>
>
>
>
> Sep 25 23:36:19 ch0012 kamailio[18696]: ERROR:  
> [core/parser/parse_from.c:53]: parse_from_header(): bad msg or missing FROM 
> header
>
> Sep 25 23:36:19 ch0012 kernel: kamailio[18696]: segfault at 0 ip 
> 004d7c4f sp 7fff16135130 error 4 in kamailio[40+42b000]
>
> Sep 25 23:36:19 ch0012 kernel: kamailio[18682]: segfault at 0 ip 
> 004d7c4f sp 7fff16135130 error 4 in kamailio[40+42b000]
>
> Sep 25 23:36:19 ch0012 kamailio[18696]: ERROR: siptrace [siptrace.c:535]: 
> sip_trace_prepare(): cannot parse FROM header
>
> Sep 25 23:36:19 ch0012 kamailio[18682]: ERROR:  
> [core/parser/parse_from.c:53]: parse_from_header(): bad msg or missing FROM 
> header
>
> Sep 25 23:36:19 ch0012 kamailio[18682]: ERROR: siptrace [siptrace.c:535]: 
> sip_trace_prepare(): cannot parse FROM header
>
> Sep 25 23:36:19 ch0012 abrt-hook-ccpp: Process 18696 (kamailio) of user 0 
> killed by SIGSEGV - ignoring (repeated crash)
>
> Sep 25 23:36:19 ch0012 abrt-hook-ccpp: Process 18682 (kamailio) of user 0 
> killed by SIGSEGV - dumping core
>
> Sep 25 23:36:20 ch0012 abrt-hook-ccpp: Saved core dump of pid 18696 to 
> core.18696 at / (2294079488 bytes)
>
> Sep 25 23:36:20 ch0012 kamailio[18759]: CRITICAL:  
> [core/pass_fd.c:277]: receive_fd(): EOF on 31
>
> Sep 25 23:36:20 ch0012 kamailio[18679]: ALERT:  [main.c:743]: 
> handle_sigs(): child process 18696 exited by a signal 11
>
> Sep 25 23:36:20 ch0012 kamailio[18679]: ALERT:  [main.c:746]: 
> handle_sigs(): core was generated
>
> Sep 25 23:36:20 ch0012 kamailio[18679]: INFO:  [main.c:768]: 
> handle_sigs(): terminating due to SIGCHLD
>
> Sep 25 23:36:20 ch0012 kamailio[18759]: INFO:  [main.c:823]: sig_usr(): 
> signal 15 received
>
> Sep 25 23:36:20 ch0012 kamailio[18754]: INFO:  [main.c:823]: sig_usr(): 
> signal 15 received
>
> Sep 25 23:36:20 ch0012 kamailio[18758]: INFO:  [main.c:823]: sig_usr(): 
> signal 15 received
>
> Sep 25 23:36:20 ch0012 kamailio[18757]: INFO:  [main.c:823]: sig_usr(): 
> signal 15 received
>
> Sep 25 23:36:20 ch0012 kamailio[18753]: INFO:  [main.c:823]: sig_usr(): 
> signal 15 received
>
> Sep 25 23:36:20 ch0012 kamailio[18752]: INFO:  [main.c:823]: sig_usr(): 
> signal 15 received
>
> Sep 25 23:36:20 ch0012 kamailio[18756]: INFO:  [main.c:823]: sig_usr(): 
> signal 15 received
>
> Sep 25 23:36:20 ch0012 kamailio[18751]: INFO:  [main.c:823]: sig_usr(): 
> signal 15 received
>
> Sep 25 23:36:20 ch0012 kamailio[18746]: INFO:  [main.c:823]: sig_usr(): 
> signal 15 received
>
> Sep 25 23:36:20 ch0012 kamailio[18747]: INFO:  [main.c:823]: sig_usr(): 
> signal 15 received
>
> Sep 25 23:36:20 ch0012 kamailio[18750]: INFO:  [main.c:823]: sig_usr(): 
> signal 15 received
>
>
>
> I have exported from the core file some info (file is attached), any idea ?
>
>
>
> BR
>
>
>
> Laurent
>
>
>
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Re: [SR-Users] recording audio stream

2018-11-17 Thread Mojtaba
In both module you need third service for recording streams.
In RTPEngine you needed rtpengine-daemon-recording. And in RTPProxy you
need extractaudio utility for extract audio streams.
With Regards.Mojtaba

On Sun, 18 Nov 2018 00:18 David Villasmil, 
wrote:

> I think he’s using rtpproxy, not rtp-engine. :)
> On Sat, 17 Nov 2018 at 20:21, Alex Balashov 
> wrote:
>
>> Hi,
>>
>> The documentation here explains fairly well how to set up the recording
>> component of RTPEngine. Recording is then enabled via a command sent in
>> the rtpengine_offer()/manage() functions.
>>
>> https://github.com/sipwise/rtpengine
>>
>> -- Alex
>>
>> On Sat, Nov 17, 2018 at 10:19:02AM -0800, b...@telopar.net wrote:
>>
>> > Hi All:
>> >
>> > I have a project that requires that I record the audio streams from an
>> ongoing phone call. It seems that rtp_proxy module should do that, but I
>> can't seem to get it to work.
>> >
>> > My config file:
>> >
>> > *if(nat_uac_test("8")) { //rtpproxy_manage("co"); xlog("L_INFO",
>> "rtpproxy
>> > Call is going to private IPv4 Media Server Engage RTPProxy Now\n");
>> > rtpproxy_manage("crwie","10.1.1.2"); } else { //rtpproxy_manage("cor");
>> > xlog("L_INFO", "rtpproxy Call is coming from a private IPv4 Media Server
>> > Engage RTPProxy Now\n"); rtpproxy_manage("crwei","10.1.1.2"); } *There
>> > appears to be conversations going on with rtpproxy on both legs of the
>> call.
>> > But I seem to be missing the step that records the audio*. *Anybody
>> have an
>> > example of how to do this.
>> >
>> > TIA, Bill
>> > **
>> >
>>
>> > ___
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>>
>>
>> --
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>>
>> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
>> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>>
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On 18 Nov 2018 00:18, "David Villasmil" 
wrote:

I think he’s using rtpproxy, not rtp-engine. :)

On Sat, 17 Nov 2018 at 20:21, Alex Balashov 
wrote:

> Hi,
>
> The documentation here explains fairly well how to set up the recording
> component of RTPEngine. Recording is then enabled via a command sent in
> the rtpengine_offer()/manage() functions.
>
> https://github.com/sipwise/rtpengine
>
> -- Alex
>
> On Sat, Nov 17, 2018 at 10:19:02AM -0800, b...@telopar.net wrote:
>
> > Hi All:
> >
> > I have a project that requires that I record the audio streams from an
> ongoing phone call. It seems that rtp_proxy module should do that, but I
> can't seem to get it to work.
> >
> > My config file:
> >
> > *if(nat_uac_test("8")) { //rtpproxy_manage("co"); xlog("L_INFO",
> "rtpproxy
> > Call is going to private IPv4 Media Server Engage RTPProxy Now\n");
> > rtpproxy_manage("crwie","10.1.1.2"); } else { //rtpproxy_manage("cor");
> > xlog("L_INFO", "rtpproxy Call is coming from a private IPv4 Media Server
> > Engage RTPProxy Now\n"); rtpproxy_manage("crwei","10.1.1.2"); } *There
> > appears to be conversations going on with rtpproxy on both legs of the
> call.
> > But I seem to be missing the step that records the audio*. *Anybody have
> an
> > example of how to do this.
> >
> > TIA, Bill
> > **
> >
>
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>
> --
> Alex Balashov | Principal | Evariste Systems LLC
>
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
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Re: [SR-Users] Logging from cfg file

2018-12-29 Thread Mojtaba
Hi,
You could use xlog instead of log, like this:
 xlog("Register request...\n");
With Regards.Mojtaba

On Sat, Dec 29, 2018 at 11:14 AM Muhammad Allaudin
 wrote:
>
> Hi everyone,
>
> I have recently started working on Kamailio, I am trying to learn the basics. 
> For "Register" route, I am trying to print the log but log is not appearing 
> in log file. Other logs are there, only this one is missing.
>
> if(is_method("REGISTER")){
>  log(2, "Register request");
> }
>
> I will be really helpful if you can help me with this.
>
> Thanks & Kind Regards
>
> Regards,
> M.Allaudin
> Software Engineer
> Nowtel Group
> Phone # +92-333-8291874
> Blogger @ allaudin.github.io
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[SR-Users] A bit trick in msg_translator.c source file

2018-12-29 Thread Mojtaba
Hello ,
Does anybody know why the lenght of both ip-address (s and name) in
check_via_address function in msg_translator.c source file dose not
check first?
Actually i think it's better do.
For example if via:192.168.122.1 and src:192.168.122.100, the result
returned true, as they are not the same!
With.Regards. Mojtaba
Happy new year

--Mojtaba Esfandiari.S

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Re: [SR-Users] Dev documentation on implementing event_route events in custom module

2019-01-18 Thread Mojtaba
Hello Ivan,
Actually i haven't seen documentation but there is a good reference for it,
the impelemention of modules. You could look around them to find out. It is
very strightforwards.


On Fri, 18 Jan 2019, 18:57 Ivan Ribakov,  wrote:

> Is there any documentation for module developers that lays out API for
> registering and triggering events from a custom Kamailio module (similar to
> https://kamailio.org/docs/modules/5.0.x/modules/tm.html#tm.event_routes)?
>
> I have been referring to
> https://www.kamailio.org/docs/kamailio-devel-guide/ up until now but it
> doesn’t seem to cover event_routes.
>
> Thanks in advance,
> Ivan
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Re: [SR-Users] Dev documentation on implementing event_route events in custom module

2019-01-18 Thread Mojtaba
I'm gone away right now. I'll send for you as soon as

On Fri, 18 Jan 2019, 22:34 Ivan Ribakov,  wrote:

> Thanks for the suggestion. I thought about that, but TM module is not one
> of the smallest. If there is a smaller, simpler model that uses events that
> someone could recommend I could look into it. But in any case, I was
> looking for something more documented to learn about all capabilities of
> the events, not just the ones that might be used by some module.
>
> On 18 Jan 2019, at 17:43, Mojtaba  wrote:
>
> Hello Ivan,
> Actually i haven't seen documentation but there is a good reference for
> it, the impelemention of modules. You could look around them to find out.
> It is very strightforwards.
>
>
> On Fri, 18 Jan 2019, 18:57 Ivan Ribakov,  wrote:
>
>> Is there any documentation for module developers that lays out API for
>> registering and triggering events from a custom Kamailio module (similar to
>> https://kamailio.org/docs/modules/5.0.x/modules/tm.html#tm.event_routes)?
>>
>> I have been referring to
>> https://www.kamailio.org/docs/kamailio-devel-guide/ up until now but it
>> doesn’t seem to cover event_routes.
>>
>> Thanks in advance,
>> Ivan
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Re: [SR-Users] Dev documentation on implementing event_route events in custom module

2019-01-18 Thread Mojtaba
Yes, as YASIN  said, you could look register module. In the following i
paste a simple example of definition:

  int route_no;

if(hi1_eventrb){
route_no = route_get(_rt, hi1_eventrb);
if (route_no == -1){
LM_ERR("Failed to hi1 route block number of \"%s\", route_get()
function is failed\n", hi1_eventrb);
return -1;
}
if (main_rt.rlist[route_no]==0){
LM_WARN("The hi1 route block \"%s\" is empty, doesn't
exist\n",hi1_eventrb);
return -1;
}
LM_DBG("hi1_eventrb is set successfully to %s route block\n",
hi1_eventrb);
hi1_eventrb_no = route_no;
}
return 1
=
And here sipmle example of calling event_route in your code:
int ret;
struct run_act_ctx ra_ctx;
struct sip_msg *msg;


init_run_actions_ctx(_ctx);

if (run_actions(_ctx, main_rt.rlist[hi1_eventrb_no], msg) < 0) {
ret = -1;
LM_DBG("Error while trying run script in route block\n");
return ret;
}
return 1;


With Regards.Mojtaba

On Fri, Jan 18, 2019 at 11:07 PM YASIN CANER  wrote:

> Hello
>
> You can look registrar module that has usrloc:contact-expire
>
> Good luck
>
> Yasin Caner
>
> Mojtaba  şunları yazdı (18 Oca 2019 22:34):
>
> I'm gone away right now. I'll send for you as soon as
>
> On Fri, 18 Jan 2019, 22:34 Ivan Ribakov,  wrote:
>
>> Thanks for the suggestion. I thought about that, but TM module is not one
>> of the smallest. If there is a smaller, simpler model that uses events that
>> someone could recommend I could look into it. But in any case, I was
>> looking for something more documented to learn about all capabilities of
>> the events, not just the ones that might be used by some module.
>>
>> On 18 Jan 2019, at 17:43, Mojtaba  wrote:
>>
>> Hello Ivan,
>> Actually i haven't seen documentation but there is a good reference for
>> it, the impelemention of modules. You could look around them to find out.
>> It is very strightforwards.
>>
>>
>> On Fri, 18 Jan 2019, 18:57 Ivan Ribakov,  wrote:
>>
>>> Is there any documentation for module developers that lays out API for
>>> registering and triggering events from a custom Kamailio module (similar to
>>> https://kamailio.org/docs/modules/5.0.x/modules/tm.html#tm.event_routes)?
>>>
>>>
>>> I have been referring to
>>> https://www.kamailio.org/docs/kamailio-devel-guide/ up until now but it
>>> doesn’t seem to cover event_routes.
>>>
>>> Thanks in advance,
>>> Ivan
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Re: [SR-Users] Additional Param Check in ConsumeCredential Auth

2019-01-20 Thread Mojtaba
Hello Prabhat,
Would you look at this function pv_www_authenticate in auth module? It
seems it would be useful for you
You could use the email as authentication parameter too,
With Regards.Mojtaba


On Mon, Jan 21, 2019 at 10:08 AM Prabhat Kumar 
wrote:

> Hi,
>
> I want to check additional param in Auth module. For. ex., other than
> username and secret , i want to validate user by its email address. How can
> i do that?
>
> Regards,
>
> Prabhat Kumar
>
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Re: [SR-Users] IMS early media

2018-12-21 Thread Mojtaba
Hello Tsvetan.
Actually you could use SIP Early media in AS and also with cscf.
If you choice the first way, i think it is very simple and strightforward
because you just use early media functions on your AS. For example in
Astrisk you could use Progress application and 'm' option in Dial
application in your dialplan.
In second way you should check in Reply-Route block,if you got 180
ringing,  you have to use rtpproxy-stream funtion for doing sip early.

Wih Regards.Mojtaba Esfandiari.S

On Fri, 21 Dec 2018, 16:34 Tsvetan Filev, 
wrote:

> Hi all.
>
> I want to use SIP early media to play music to the caller in kamailio
> IMS installation like this:
> http://www.sharetechnote.com/html/IMS_SIP_EarlyMedia.html
>
> I looked a little bit but didn't find ready solution. The information is
> vague on this topic.
> Should this be done through a module or application server ?
> May I need to handle ringing in onreply_route and send OK with SDP to
> the caller in SCSCF ?
>
> Regards.
>
>
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Re: [SR-Users] P-CSCF without Rx interface

2018-12-10 Thread Mojtaba
Hi,
The answer is right,

About your question: "I'm not able to find how to tell P-CSCF where to
direct the sip traffic. It obviously should happen in
route[REGISTER]."

In [REGISTER] route block you could see this line:
$ru = "sip:"+$rU+"@"+$rd;
Actually you are setting R-URI, In this line the $rd PV refer to ICSCF
in IMS network.
With Regards. Mojtaba
On Mon, Dec 10, 2018 at 6:15 PM Iman Mohammadi
 wrote:
>
> Hi
>
> You can comment RX in pcscf.cfg but if you don't use a PCRF or a EPC that's 
> not VoLTE call any more and You can use a softphone like Zoiper to set up 
> call on your ims and check your ports that you specify for P-CSCF and I-CSCF 
> in DNS and cfg files for solving loop problem
>
> On Mon, Dec 10, 2018, 17:53 Tsvetan Filev  wrote:
>>
>> Hi All.
>>
>> May be this has been asked before but I was not able to find an answer.
>> I setup IMS with 3 instances of kamailio: P-CSCF, I-CSCF and S-CSCF (I
>> got the config files from here
>> https://github.com/kamailio/kamailio/tree/master/misc/examples/ims).
>> I would like to register two clients to the IMS and make a VoLTE like call.
>> So I want to setup my P-CSCF to send SIP messages to my I-CSCF without
>> Rx interface (without using PCRF).
>> I'm not able to find how to tell P-CSCF where to direct the sip traffic.
>> It obviously should happen in route[REGISTER].
>> Right now P-CSCF sends all REGISTER requests to itself and goes to a loop.
>>
>> Regards.
>>
>>
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Re: [SR-Users] is_e164 logic to detect valid number

2018-12-02 Thread Mojtaba
Hello,
it's strange that why this function (actually i mean e164_check
function) doesn’t check the length of number!
Based on Recommendation ITU-T E.164, Annex A.3 Number length, it
should be matched with one of structures that is explained there
So you could do check it yourself with regex.
With Regards. Mojtaba
On Mon, Dec 3, 2018 at 8:39 AM Patrick Wakano  wrote:
>
> Hello list,
> Hope you are all doing fine!
>
> I am using the is_e164() function to validate the number we receive, and I 
> come to see that the number +555 was accepted
> After some googling it looks like(it is not very clear though) that 7 digits 
> are the minimum we could have for e164 numbers but after checking the source 
> code, I saw it accepts anything starting with + and having between 2 and 16 
> numbers. So is it really valid to have a number with just 2 digits? What is 
> the case?
>
> Thank you,
> Kind regards,
> Patrick Wakano
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Re: [SR-Users] database connections per child process

2018-11-22 Thread Mojtaba
Actually No.
Actually the connection to database is done in fork process. So if you
have two fork process with one sqlcon param, You'll have two TCP
connection.
If you add one another sqlcon param in modparam, you will have four
tcp connection. at finally you will have:
(Number_Of_Connection=Fork_no * sqlcon_no)
It is the concept of implementing Kamailio project. So if you want
another treatment in this regards, you have to change in PROC_INIT
process in child_init function.
With Regards.Mojtaba

On Thu, Nov 22, 2018 at 7:32 AM Karthik Srinivasan
 wrote:
>
> Hello,
>
> Is there a formula/algorithm that determines how many TCP connections are 
> made per child process to a database server defined by sqlops' sqlcon 
> parameter?
>
> ex: if i have 1 database on a database server defined by sqlcon, and i have 1 
> UDP child process, i notice that the child process has made 2 TCP connections 
> to the database server.
> ex: if i have 3 databases on the same database server defined by 3 sqlcon 
> statements (1 pertaining to each database) and i have 4 UDP child processes, 
> i notice that each child process makes 4 TCP connections to the database 
> server.
> ex: if i have 3 databases on the same database server defined by 3 sqlcon 
> statements (1 pertaining to each database) and i have 6 TCP child processes, 
> i notice that each child process makes 4 TCP connections to the database 
> server.
>
> Basically wanting to know what governs the number of tcp connections a child 
> process makes to a particular database server.
>
> Thank you,
>
> Karthik
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Re: [SR-Users] SIP Express Media Server (aka SEMS)

2019-01-09 Thread Mojtaba
Hello,
I've never heard that SEMS Support video, But I just know if you switch
enable_rtprelay's option off, It would  works as p2p.
So using these feature of SEMS is optional and you could keep using SEMS
with no worry!


On Wed, Jan 9, 2019 at 12:35 PM Abhijit Dutta  wrote:

> Hi All,
>
> Is the development work for SEMS server still on or stopped? Does it
> support video?
>
> Thanks
> Abhijit
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[SR-Users] siptarce module and SIP-Replying Messages

2019-01-14 Thread Mojtaba
Hello,
I am using siptrace module in Kamailio. As you know, it sends a copy of
signalling message for specific destination. Like HOMER project.
Because of  i won't run main route block in destination, i just  have
changed this block like below:
request_route {
//do some works.
drop();
exit;
}
I know it would make a question in your mind why i do this? No problem,
Let's come back to question. All things works correct except Replying SIP
message. The destination is trying to relay Replying SIP message that it
get from siptrace module in other server.
How can i disable replying SIP mesages?
With Best Regards
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Re: [SR-Users] TOPOH module im REGISTER method.

2018-09-12 Thread Mojtaba
Hi all,
Any update please?
Thanks with regards.Mojtaba
On Wed, Sep 5, 2018 at 12:19 PM Mojtaba  wrote:
>
> Hi all,
> I try to used SEMS in this regards, But i have this issue.
> https://lists.kamailio.org/pipermail/sr-users/2018-August/102753.html
> Any updates?
> Thanks
> On Tue, Sep 4, 2018 at 2:07 PM Mojtaba  wrote:
> >
> > Hello,
> > I am used topoh module in kamailio, Why this module does not work (No
> > change ) in Register request Method?
> > I need to change UA' ip address in Via header in Register request
> > before forwarded to my network? Does the topoh module used in this
> > situation?
> > I have not this issue in INVITE request, I just have this issue in
> > Register request.
> > Thanks.
> > With Regards.Mojtaba
> >
> > --
> > --Mojtaba Esfandiari.S
>
>
>
> --
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Re: [SR-Users] Record Route Parameters

2018-09-15 Thread Mojtaba
Hi,
As Alex mentioned, "they will not be changed by comaliance UA", But be
notice that if you want to relay your request to other network, and
these is a SBC on upstream networks, These parameters could be removed
by them, As in your Kamailio as SBC, you remove all record_route, and
add its IP as record_route.

remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE")) {
   record_route();
}
With Regards.Mojtaba
On Fri, Sep 14, 2018 at 6:24 PM Alex Balashov  wrote:
>
> They will not be changed by compliance user agents. In that sense it's safe. 
> However, be mindful of the fact that these parameters can be intercepted or 
> altered by a malicious user agent, and that adding them in multiplicity 
> contributes to the overall size of the message.
>
> --
> Sent from mobile. Apologies for brevity and errors.
>
> -Original Message-
> From: Duarte Rocha 
> To: sr-users@lists.kamailio.org
> Sent: Fri, 14 Sep 2018 9:46 AM
> Subject: [SR-Users] Record Route Parameters
>
> Greetings,
>
> I have Kamailio as an SBC with mutiple IP's and I need to save some
> information in the first Invite about the routing made internally in the
> Kamailio.
>
> I'm thinking about adding some parameters to Record-Route and retrieve them
> in the Route header. Is this safe ? Can I be sure that those parameters
> won't be deleted or changed?
>
> Best Regards,
>
> Duarte Rocha
>
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Re: [SR-Users] Record-Route Parameters

2018-09-15 Thread Mojtaba
Let me know do you need to add all IP's of kamailio in Record_Routes ?
of course you need it when using cross-protocol forwarding.
If you needed, you should add them by re-writing headers directly,
else you could work with "enable_double_rr" parameter, to add one
Record_Route in header.
With Regards.Mojtaba
On Fri, Sep 14, 2018 at 9:25 PM Duarte Rocha  wrote:
>
> Greetings,
>
> My Kamailio has 2 IP's and, as such, it will generate two Record-Routes when 
> the Request is relayed.
>
> I want to add one parameter to one of the Record Routes and another parameter 
> to the other Record-Route. However, if i use add_rr_param() it will 
> automatically add the parameters to the two Record-Route Headers.
>
> Is there a way to do what i want without me re-writing the headers directly ?
>
> Cheers,
>
> Duarte Rocha
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Re: [SR-Users] Allow registration only for one client per AOR

2018-09-15 Thread Mojtaba
Hi, Mybe it is better to add some policy in kamailio config file to do it.
As you could see in kamailio.cfg,
   if (!impu_registered("location")) {
xlog("L_ERR", "Not REGISTERED\n");
save("PRE_REG_SAR_REPLY","location");
exit;
} else {
isc_match_filter_reg("1","location");
save("REG_SAR_REPLY","location");
exit;
}
If the IMPU is registered or not, the save function is executed, You
could add your favourite policy here,
WIth Regards.Mojtaba
On Sat, Sep 15, 2018 at 1:02 AM Henning Westerholt  wrote:
>
> Am Freitag, 14. September 2018, 16:03:33 CEST schrieb eyas barhouk:
> > Dear thanks for your response
> > I'm using kamilio ims_registrar-scscf module Instead of register module
> >
> > So is the parameter max_contacts on thats module do the same ??
>
> Hello,
>
> it seems it does. I am not an expert for this IMS modules. It seems that the
> storage for the registrar is also the ims_usrloc_scscf module. But maybe
> somebody which uses this modules can comment on that too.
>
> This module unfortunately has no documentation, I will open a github issue for
> it. It seems that the parameter is named "maxcontact". There is another
> parameter "maxcontact_behaviour", set to 1 it will reject the contact, set to
> 2 it will replace the oldest.
>
> Best regards,
>
> Henning
>
> --
> Henning Westerholt
> https://skalatan.de/blog/
>
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Re: [SR-Users] kamailio + rtpengine + custom filename

2018-09-12 Thread Mojtaba
It is pretty straightforward, There is simple way to use it.
You could use exec module to run shell command in kamailio,
for example:
if (is_method("BYE")) {
   exec_msg("mv /var/lib/rtpengine-recording/$ci*
/your_custom_path/new_filename.wav");
}
When the BYE method for specific CALL_ID is received, The exec command
should be run, It mv the current recorded file to special path with
new filename.
Notice that the default path of recording file in rtpengine has not
sufficient permossion, Then you should change the directory permission
at first. like this:
cmod 755 /var/lib/rtpengine-recording/ -R

With Regards.Mojtaba
On Tue, Sep 11, 2018 at 3:28 PM Жан Базаров  wrote:
>
> Message-ID:
> 
> Content-Type: text/plain; charset="UTF-8"
>
> Hello,
> You could use metadata to save your favourite filename on it. After
> recording is finished, replace it. The metadata file is created based
> on  for each call.
> With Regards.
> Mojtaba
> On Mon, Sep 10, 2018 at 8:53 PM Жан Базаров  wrote:
> >
> > Hello! Filenames have the format --.
> >  How i can change filename in start_recording() ???
>
> Thx you. Can you help me with it task? Show me, please, an example. I want 
> filename  of the following format, for example: 
> 'output-dir//mm/dd/fromnumber/callid-tonumber-uniqueid.wav'.
>
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Re: [SR-Users] re- double record route

2018-09-17 Thread Mojtaba
Your response confused me, But let me describe some rules.
The BYE request is used the record route header in it's INVITE request
in Dialogs. Then when UE want to send BYE, It add a all record route
in BYE message.
On other hand, the SIP stack that UE is used, may be different, So you
must be sure about it. You have two choice: Diving to SIP stack in UE
(doesn't good choice), and use the power of Kamailio.
For example, Use Path header to save the second record route (or your
favourite record route) in INVITE message, then if the UE send BYE
message which the Path header in, use this Path header for relaying in
kamailio.
As i said, It is needed to paste the log for better understanding.
With Regards.Mojtaba
On Mon, Sep 17, 2018 at 2:17 AM Wilkins, Steve  wrote:
>
> Hi Mojtaba,
>
> But when I send the 'BYE' doesn't the double Record-Route from the 'INVITE' 
> (from IOS) need to be there, so that IOS can find it's Proxies and complete 
> the transaction and send back a '200 OK'?
>
> Thank you,
>
> -Original Message-
> From: sr-users  On Behalf Of Mojtaba
> Sent: Sunday, September 16, 2018 12:47 PM
> To: Kamailio (SER) - Users Mailing List 
> Subject: Re: [SR-Users] re- double record route
>
> What do you mean exactly? Do you mean the INVITE is received from IOS has 
> double record route? If this true, the Kamailio remove all record route when 
> the INVITE request is received by default. In other words, Kamailio remove 
> all record routes form downstream or upstream in INVITE request by default.
> You should paste a simple wireshark to solve it as soon.
> With Regards.Mojtaba
> On Sun, Sep 16, 2018 at 6:10 PM Wilkins, Steve  wrote:
> >
> > Hi Henning,
> >
> >
> >
> > Yes I do have that enabled.  What is happening is that one of the providers 
> > on IOS is sending a double record route on the INVITE, but it is getting 
> > lost somewhere so when I send a 'BYE', I get a "404 not here".  When I look 
> > at the SIP message I see only one of the record routes from the INVITE, 
> > therefore my assumption is, this is what is causing the 404, because 
> > everything else looks good.
> >
> >
> >
> > _______
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Re: [SR-Users] Kamailio Peer Trunk in Asterisk

2019-03-13 Thread Mojtaba
Hello,
In regard of register Kamailio to asterisk, You have to use uac module, But
Why do you choice this!
It's better to keep on your way with IP Authentication even if both server
are not in the same network,too. Because Kamailio could work multi-homed.
Just Enable mhome=1 in kamailio.
With Best Regards.Mojtaba


On Wed, Mar 13, 2019 at 10:33 AM Isravel Raja Thangamani <
ir...@ebusinessbpo.com> wrote:

> Hi,
>
> I wanted to achieve the following scenario with Kamailio and
> Asterisk/FreeSwitch
>
> Kamailio as a trunk for Receiving incoming calls and forward to Asterisk
> and Make outgoing calls. My current working setup like this
>
>
> VoIP Provider -> Kamailio -> Asterisk Clusters
>
> Where Kamailio(NAT for public) and Asterisk in the same local network, So
> I had to do only one thing just create the SIP trunk in Asterisk for
> authenticating all the Kamailio Incoming calls and making outgoing calls
> like this
>
> [kamailio]
> host=172.18.1.1
> type=friend
> disallow=all
> allow=ulaw
> context=trunkinbound
>
> This is IP based authentication, This is working fine since both servers
> are in the same local network or having the public IP's
>
> What I want is, I want to register the Kamailio Trunk in an asterisk with
> Username Password authentication,
>
> My current setup for making that,
>
> VoIP Provider -> Kamailio(Public) -> Asterisk(Randomly Changing Public IP,
> Broadband Connection)
>
> I want to register Kamailio SIP in asterisk trunk so that I can forward
> the incoming calls to asterisk via SIP extension
>
> [kamailio]
> host=172.18.1.1
> user=user
> secret=password
> type=friend
> disallow=all
> allow=ulaw
> context=trunkinbound
>
> Let me know if anyone doesn't understand my concern
>
>
>
> Thanks & Regards,
>
>
> Isravel Raja N. Thangamani
>
> Work US: +1 424 354 8443 | Home US: +1 480 588 2008
> Mobile PH: +63 998 596 7805 | Off Ext: 7901
> Address: 15th Fl., Cebu IT Tower - 1, Cebu Business Park, Cebu City,
> Philippines 6000
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>
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Re: [SR-Users] cdp avps issues in kamailio 5.2

2019-03-13 Thread Mojtaba
Hello,
The CER/CEA messages are used just for connections between two peers
diameter interface. Did you use FHOSS or IMS_DIAMETER_SERVER module? In
both cases, you should not have any issue. Anyway what mandatory parameters
do you mean exactly?
WIth Best Regards. Mojtava
<https://www.kamailio.org/docs/modules/5.1.x/modules/ims_diameter_server.html>

On Tue, Mar 12, 2019 at 8:17 PM Pafel  wrote:

> Hello,
>
> I am testing ims setup with kamailio 5.2 using the default configuration
> provided here:
> https://github.com/kamailio/kamailio/tree/master/misc/examples/ims/ .
> Тhere is an issue I am facing with cdp avp in request sent to hss. There
> are some missing mandatory parameters in capabilities exchange request like
> 257 host-ip-address for example. Any idea how to add it via icscf.xml or
> kamailio.cfg ? The same parameter is passed to hss with kamailio 5.1 using
> default ims configuration. I read cdp and cdp_avp documentation but
> couldn't understand if it is possible to add it.
>
> Regards,
> Pavel Siderov
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Re: [SR-Users] cdp avps issues in kamailio 5.2

2019-03-13 Thread Mojtaba
The sound is strange, Actually the host-IP-Addres should not be missed in
new version because it is mandatory option,
I guess it would be nice to check it with latest version again.
With Regards. Mojtaba

On Wed, Mar 13, 2019 at 2:07 PM Pavel S  wrote:

> Hello Mojtava,
>
> > Did you use FHOSS or IMS_DIAMETER_SERVER module?
> No, I am using our own hss server.
>
> >Anyway what mandatory parameters do you mean exactly?
> According to RFC6733 section 5.3.1 - 
> https://tools.ietf.org/html/rfc6733#section-5.3.1
>
>The Capabilities-Exchange-Request (CER), indicated by the Command
>Code set to 257 and the Command Flags' 'R' bit set, is sent to
>exchange local capabilities.  Upon detection of a transport failure,
>this message MUST NOT be sent to an alternate peer.
>
>When Diameter is run over SCTP [RFC4960] or DTLS/SCTP [RFC6083],
>which allow for connections to span multiple interfaces and multiple
>IP addresses, the Capabilities-Exchange-Request message MUST contain
>one Host-IP-Address AVP for each potential IP address that MAY be
>locally used when transmitting Diameter messages.
>
>   Message Format
>
>   ::= < Diameter Header: 257, REQ >
>{ Origin-Host }
>{ Origin-Realm }
> 1* { Host-IP-Address }
>{ Vendor-Id }
>{ Product-Name }
>[ Origin-State-Id ]
>  * [ Supported-Vendor-Id ]
>  * [ Auth-Application-Id ]
>  * [ Inband-Security-Id ]
>  * [ Acct-Application-Id ]
>  * [ Vendor-Specific-Application-Id ]
>[ Firmware-Revision ]
>  * [ AVP ]
>
> Kamailio 5.1 with loaded cdp, cdr_avp, ims_icsf modules and default 
> configuration described here 
> https://github.com/kamailio/kamailio/tree/master/misc/examples/ims/icscf 
> sends Host-IP-Address but Kamailio 5.2 with the same configuration doesn't.
>
> Any idea how to fix it?
>
> Regards,
>
> Pavel Siderov
>
>
> >Hello,
> >The CER/CEA messages are used just for connections between two peers
> >diameter interface. Did you use FHOSS or IMS_DIAMETER_SERVER module? In
> >both cases, you should not have any issue. Anyway what mandatory parameters
> >do you mean exactly?
> >WIth Best Regards. Mojtava
> ><https://www.kamailio.org/docs/modules/5.1.x/modules/ims_diameter_server.html>
>
> >On Tue, Mar 12, 2019 at 8:17 PM Pafel  ><https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> wrote:
> >
> >>* Hello,
> *>>>>* I am testing ims setup with kamailio 5.2 using the default 
> configuration
> *>>* provided here:
> *>>* https://github.com/kamailio/kamailio/tree/master/misc/examples/ims/ 
> <https://github.com/kamailio/kamailio/tree/master/misc/examples/ims/> .
> *>>* Тhere is an issue I am facing with cdp avp in request sent to hss. There
> *>>* are some missing mandatory parameters in capabilities exchange request 
> like
> *>*> 257 host-ip-address for example. Any idea how to add it via icscf.xml or
> *>*> kamailio.cfg ? The same parameter is passed to hss with kamailio 5.1 
> using
> *>*> default ims configuration. I read cdp and cdp_avp documentation but
> *>*> couldn't understand if it is possible to add it.
> *>>>>* Regards,
> *>>* Pavel Siderov
> *>>* ___
> *>>* Kamailio (SER) - Users Mailing List
> *>>* sr-users at lists.kamailio.org 
> <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
> *>*> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users 
> <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
> *>>
>
> ___
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> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>


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Re: [SR-Users] ICSCF and SCSCF Problem

2019-03-11 Thread Mojtaba
): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 154, column 61: Can't set module
> parameter
>  0(7707) ERROR:  [modparam.c:150]: set_mod_param_regex(): No module
> matching  found
>  0(7707) :  [cfg.y:3435]: yyerror_at(): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 161, column 51: Can't set module
> parameter
>  0(7707) ERROR:  [modparam.c:150]: set_mod_param_regex(): No module
> matching  found
>  0(7707) :  [cfg.y:3435]: yyerror_at(): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 165, column 53: Can't set module
> parameter
>  0(7707) ERROR:  [modparam.c:150]: set_mod_param_regex(): No module
> matching  found
>  0(7707) :  [cfg.y:3435]: yyerror_at(): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 170, column 51: Can't set module
> parameter
>  0(7707) ERROR:  [cfg.y:3295]: yyparse(): cfg. parser: failed to
> find command I_scscf_select (params 1)
>  0(7707) :  [cfg.y:3435]: yyerror_at(): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 325, column 24: unknown command,
> missing loadmodule?
>
>  0(7707) ERROR:  [cfg.y:3295]: yyparse(): cfg. parser: failed to
> find command I_scscf_drop (params 0)
>  0(7707) :  [cfg.y:3435]: yyerror_at(): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 337, column 16: unknown command,
> missing loadmodule?
>
>  0(7707) ERROR:  [cfg.y:3295]: yyparse(): cfg. parser: failed to
> find command I_perform_user_authorization_request (params 2)
>  0(7707) :  [cfg.y:3435]: yyerror_at(): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 339, column 73: unknown command,
> missing loadmodule?
>
>  0(7707) ERROR:  [cfg.y:3295]: yyparse(): cfg. parser: failed to
> find command I_scscf_select (params 1)
>  0(7707) :  [cfg.y:3435]: yyerror_at(): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 350, column 43: unknown command,
> missing loadmodule?
>
>  0(7707) ERROR:  [cfg.y:3295]: yyparse(): cfg. parser: failed to
> find command I_scscf_drop (params 0)
>  0(7707) :  [cfg.y:3435]: yyerror_at(): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 358, column 42: unknown command,
> missing loadmodule?
>
>  0(7707) ERROR:  [cfg.y:3295]: yyparse(): cfg. parser: failed to
> find command I_scscf_drop (params 0)
>  0(7707) :  [cfg.y:3435]: yyerror_at(): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 386, column 17: unknown command,
> missing loadmodule?
>
>  0(7707) ERROR:  [cfg.y:3295]: yyparse(): cfg. parser: failed to
> find command I_scscf_select (params 1)
>  0(7707) :  [cfg.y:3435]: yyerror_at(): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 400, column 25: unknown command,
> missing loadmodule?
>
>  0(7707) ERROR:  [cfg.y:3295]: yyparse(): cfg. parser: failed to
> find command I_scscf_drop (params 0)
>  0(7707) :  [cfg.y:3435]: yyerror_at(): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 415, column 17: unknown command,
> missing loadmodule?
>
>  0(7707) ERROR:  [cfg.y:3295]: yyparse(): cfg. parser: failed to
> find command I_perform_location_information_request (params 2)
>  0(7707) :  [cfg.y:3435]: yyerror_at(): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 428, column 57: unknown command,
> missing loadmodule?
>
>  0(7707) ERROR:  [cfg.y:3295]: yyparse(): cfg. parser: failed to
> find command I_scscf_select (params 1)
>  0(7707) :  [cfg.y:3435]: yyerror_at(): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 433, column 25: unknown command,
> missing loadmodule?
>
>  0(7707) ERROR:  [cfg.y:3295]: yyparse(): cfg. parser: failed to
> find command I_scscf_drop (params 0)
>  0(7707) :  [cfg.y:3435]: yyerror_at(): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 444, column 17: unknown command,
> missing loadmodule?
>
>  0(7707) ERROR:  [cfg.y:3295]: yyparse(): cfg. parser: failed to
> find command I_scscf_drop (params 0)
>  0(7707) :  [cfg.y:3435]: yyerror_at(): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 463, column 16: unknown command,
> missing loadmodule?
>
>  0(7707) ERROR:  [cfg.y:3295]: yyparse(): cfg. parser: failed to
> find command I_scscf_select (params 1)
>  0(7707) :  [cfg.y:3435]: yyerror_at(): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 476, column 25: unknown command,
> missing loadmodule?
>
>  0(7707) ERROR:  [cfg.y:3295]: yyparse(): cfg. parser: failed to
> find command I_scscf_drop (params 0)
>  0(7707) :  [cfg.y:3435]: yyerror_at(): parse error in config file
> /etc/kamailio/icscf/kamailio.cfg, line 490, column 16: unknown command

Re: [SR-Users] rat-proxy issue

2019-02-09 Thread Mojtaba
Hello,
Are you sure run RTPproxy service is run as Unix socket or not? This ERROR
in behalf of Kamailio says there is no Unix socket running!
You could set rtpproxy_sock instance in kamailio to UNIX socket or
IPV4/IPV6 UPD socket. Also you could ran RTPProxy service just as
rtpproxy_sock instance in kamailio has done.
With Best Regards.Mojtaba



On Sat, Feb 9, 2019 at 3:21 PM aroosa satti  wrote:

> Hi! I have issue regarding my rtpproxy.I have start it but it does't work
> and give error like
>
> Feb  9 02:46:50 ns1 ./kamailio[3682]: NOTICE:  [main.c:692]:
> handle_sigs(): Thank you for flying kamailio!!!
> Feb  9 02:46:53 ns1 ./kamailio[4318]: ERROR:  [main.c:1165]:
> parse_phostport(): bad protocol in unix:/var/run/rtpproxy.sock
> these are the issue I will be grateful for you if you solve this problem.
>
> kind regards
> Aroosa Satti
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>


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Re: [SR-Users] Altering received parameter in Contact

2019-01-28 Thread Mojtaba
Hello David,
Maybe this would be useful.
https://lists.kamailio.org/pipermail/sr-users/2015-November/090595.html

With best regards.Mojtaba

On Tue, 29 Jan 2019, 06:00 David Cunningham, 
wrote:

> Hello,
>
> I'm hoping someone can give advice on altering the received parameter in
> the Contact header on a reply. The situation is that a customer is
> REGISTERing using TLS, and our Kamailio replies with a 200 OK that includes
> a Contact header like this:
>
> Contact:  :5061;transport=tls>;expires=3600;received="sip:xx.xx.102.10:36051;transport=TLS"
>
> The customer's device apparently had an issue with this, and the device
> manufacturer advises that they need the ";transport=TLS" on the received
> parameter removed. This sounds like a strange requirement, but it's what
> we've been told. Does anyone know how to alter this in a Kamailio
> configuration?
>
> Our system uses www_authorize to authenticate and is running Kamailio
> 4.2.1.
>
> Thanks in advance for any help!
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
> ___
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Re: [SR-Users] IMS early media

2019-01-31 Thread Mojtaba
Great,
Using the progress in dialplan is good sounds.
 With Regards.Mojtaba

On Thu, 31 Jan 2019, 16:34 Tsvetan Filev, 
wrote:

> Hi Mojtaba.
>
> I managed to get it working in the following way:
>
> 1. I set FilterCriteria for INVITE in the user profile
> 2. In asterisk sip.conf I set outboundproxy (no need to modify DNS)
> 3. I set a new class in musiconhold.conf
> 4. I set dial plan in extensions.conf
>
> exten => 97255102,1,Progress()
> exten => 97255102,n,Dial(
> SIP/972551000...@mnc001.mcc001.3gppnetwork.org,20,m(mymoh));
>
> Tnx.
> On 28.01.19 г. 12:29 ч., Mojtaba wrote:
>
> In another way, you could don't change this file, instead of change your
> dial plan like below:
> exten => 97255102,1,Dial(
> SIP/972551000...@icscf.mnc001.mcc001.3gppnetwork.org,20
> );
> WIth Regards.Mojtaba
>
> On Mon, Jan 28, 2019 at 1:56 PM Mojtaba  wrote:
>
>> It would be like these lines with afew changes:
>> mnc001.mcc001.3gppnetwork.org.  1D IN A   10.82.10.56
>> mnc001.mcc001.3gppnetwork.org.  1D IN NAPTR 10 50 "s"
>> "SIP+D2U"""_sip._udp
>> mnc001.mcc001.3gppnetwork.org.  1D IN NAPTR 20 50 "s"
>> "SIP+D2T"""_sip._tcp
>>
>>
>> On Mon, Jan 28, 2019 at 1:37 PM Tsvetan Filev <
>> tsvetan.fi...@inno-networks.com> wrote:
>>
>>> Here is my current zone file:
>>>
>>> $ORIGIN mnc001.mcc001.3gppnetwork.org.
>>> $TTL 1W
>>> @   1D IN SOA   localhost. root.localhost. (
>>> 1   ; serial
>>> 3H  ; refresh
>>> 15M ; retry
>>> 1W  ; expiry
>>> 1D ); minimum
>>>
>>> 1D IN NSns
>>> ns  1D IN A 10.82.10.56
>>>
>>> pcscf   1D IN A 10.82.10.56
>>> _sip._udp.pcscf 1D SRV 0 0 5060 pcscf
>>> _sip._tcp.pcscf 1D SRV 0 0 5060 pcscf
>>>
>>> icscf   1D IN A 10.82.10.56
>>> _sip._udp   1D SRV 0 0 4060 icscf
>>> _sip._tcp   1D SRV 0 0 4060 icscf
>>> _sip._udp.ims   1D SRV 0 0 4060 icscf
>>> _sip._tcp.ims   1D SRV 0 0 4060 icscf
>>>
>>> scscf   1D IN A     10.82.10.56
>>> _sip._udp.scscf 1D SRV 0 0 6060 scscf
>>> _sip._tcp.scscf 1D SRV 0 0 6060 scscf
>>>
>>> as  1D IN A 10.82.10.56
>>> _sip._udp.as1D SRV 0 0 5062 as
>>> _sip._tcp.as1D SRV 0 0 5062 as
>>>
>>> hss 1D IN A 10.82.10.56
>>>
>>>
>>> How do I modify it in order to make this work ?
>>>
>>> Tnx.
>>>
>>> On 28.01.19 г. 11:50 ч., Mojtaba wrote:
>>>
>>> Hi Tsvetan,
>>> Why do you send call back to S-CSCF? You should send call back to
>>> I-CSCF.  Actually in resolve of domain "mnc001.mcc001.3gppnetwork.org"
>>> , The ICSCF's IP
>>> should be returned.
>>>  Make sure entry SRV recordd in DNS server are true.
>>> This kind of call back to IMS is true, But make sure you won't have any
>>> issue in DNS resolve.
>>>   exten => 97255102,1,Dial(
>>> SIP/972551000...@mnc001.mcc001.3gppnetwork.org,20);
>>>
>>> With Regards.Mojtaba
>>> On Mon, Jan 28, 2019 at 12:34 PM Tsvetan Filev <
>>> tsvetan.fi...@inno-networks.com> wrote:
>>>
>>>> Hi Mojtaba.
>>>>
>>>> I implemented the AS way and was able to play sound to the caller but
>>>> In order to continue the call and send the invite to SCSCF I need to use
>>>> proxy in the Dial application which is a problem (Asterisk is B2BUA not a
>>>> proxy).
>>>> I found this old question here
>>>> https://community.asterisk.org/t/how-can-i-configure-asterisk-to-act-like-a-sip-proxy-server/18464
>>>> that describes exactly the same issue.
>>>> Here is my dial plan:
>>>>
>>>> exten => 97255102,1,Progress()
>>>> exten => 97255102,n,Playback(vm-starmain, noanswer)
>>>> exten => 97255102,n,Wait(3)
>>>> exten => 97255102,n,H

Re: [SR-Users] IMS early media

2019-01-28 Thread Mojtaba
It would be like these lines with afew changes:
mnc001.mcc001.3gppnetwork.org.  1D IN A   10.82.10.56
mnc001.mcc001.3gppnetwork.org.  1D IN NAPTR 10 50 "s" "SIP+D2U"
""_sip._udp
mnc001.mcc001.3gppnetwork.org.  1D IN NAPTR 20 50 "s" "SIP+D2T"
""_sip._tcp


On Mon, Jan 28, 2019 at 1:37 PM Tsvetan Filev <
tsvetan.fi...@inno-networks.com> wrote:

> Here is my current zone file:
>
> $ORIGIN mnc001.mcc001.3gppnetwork.org.
> $TTL 1W
> @   1D IN SOA   localhost. root.localhost. (
> 1   ; serial
> 3H  ; refresh
> 15M ; retry
> 1W  ; expiry
> 1D ); minimum
>
> 1D IN NSns
> ns  1D IN A 10.82.10.56
>
> pcscf   1D IN A 10.82.10.56
> _sip._udp.pcscf 1D SRV 0 0 5060 pcscf
> _sip._tcp.pcscf 1D SRV 0 0 5060 pcscf
>
> icscf   1D IN A 10.82.10.56
> _sip._udp   1D SRV 0 0 4060 icscf
> _sip._tcp   1D SRV 0 0 4060 icscf
> _sip._udp.ims   1D SRV 0 0 4060 icscf
> _sip._tcp.ims   1D SRV 0 0 4060 icscf
>
> scscf   1D IN A 10.82.10.56
> _sip._udp.scscf 1D SRV 0 0 6060 scscf
> _sip._tcp.scscf 1D SRV 0 0 6060 scscf
>
> as  1D IN A 10.82.10.56
> _sip._udp.as1D SRV 0 0 5062 as
> _sip._tcp.as1D SRV 0 0 5062 as
>
> hss 1D IN A 10.82.10.56
>
>
> How do I modify it in order to make this work ?
>
> Tnx.
>
> On 28.01.19 г. 11:50 ч., Mojtaba wrote:
>
> Hi Tsvetan,
> Why do you send call back to S-CSCF? You should send call back to I-CSCF.
> Actually in resolve of domain "mnc001.mcc001.3gppnetwork.org"
> , The ICSCF's IP
> should be returned.
>  Make sure entry SRV recordd in DNS server are true.
> This kind of call back to IMS is true, But make sure you won't have any
> issue in DNS resolve.
>   exten => 97255102,1,Dial(
> SIP/972551000...@mnc001.mcc001.3gppnetwork.org,20);
>
> With Regards.Mojtaba
> On Mon, Jan 28, 2019 at 12:34 PM Tsvetan Filev <
> tsvetan.fi...@inno-networks.com> wrote:
>
>> Hi Mojtaba.
>>
>> I implemented the AS way and was able to play sound to the caller but In
>> order to continue the call and send the invite to SCSCF I need to use proxy
>> in the Dial application which is a problem (Asterisk is B2BUA not a proxy).
>> I found this old question here
>> https://community.asterisk.org/t/how-can-i-configure-asterisk-to-act-like-a-sip-proxy-server/18464
>> that describes exactly the same issue.
>> Here is my dial plan:
>>
>> exten => 97255102,1,Progress()
>> exten => 97255102,n,Playback(vm-starmain, noanswer)
>> exten => 97255102,n,Wait(3)
>> exten => 97255102,n,Hangup()
>>   ; This will send the call to the pcscf again
>>   ;  exten => 97255102,1,Dial(
>> SIP/972551000...@mnc001.mcc001.3gppnetwork.org,20);
>>   ; This will send the call to scscf but it will be rejected as domain
>> not supported
>>   ;  exten => 97255102,1,Dial(
>> SIP/972551000...@scscf.mnc001.mcc001.3gppnetwork.org,20);
>>
>> Can I use kamailio as an AS and implement the same ?
>>
>> Regards.
>> On 22.12.18 г. 0:06 ч., Mojtaba wrote:
>>
>> Hello Tsvetan.
>> Actually you could use SIP Early media in AS and also with cscf.
>> If you choice the first way, i think it is very simple and strightforward
>> because you just use early media functions on your AS. For example in
>> Astrisk you could use Progress application and 'm' option in Dial
>> application in your dialplan.
>> In second way you should check in Reply-Route block,if you got 180
>> ringing,  you have to use rtpproxy-stream funtion for doing sip early.
>>
>> Wih Regards.Mojtaba Esfandiari.S
>>
>> On Fri, 21 Dec 2018, 16:34 Tsvetan Filev, <
>> tsvetan.fi...@inno-networks.com> wrote:
>>
>>> Hi all.
>>>
>>> I want to use SIP early media to play music to the caller in kamailio
>>> IMS installation like this:
>>> http://www.sharetechnote.com/html/IMS_SIP_EarlyMedia.html
>>>
>>> I looked a little bit but didn't find ready solution. The information is
>>> vague on this topic.
>>&

Re: [SR-Users] IMS early media

2019-01-28 Thread Mojtaba
In another way, you could don't change this file, instead of change your
dial plan like below:
exten => 97255102,1,Dial(
SIP/972551000...@icscf.mnc001.mcc001.3gppnetwork.org,20
);
WIth Regards.Mojtaba

On Mon, Jan 28, 2019 at 1:56 PM Mojtaba  wrote:

> It would be like these lines with afew changes:
> mnc001.mcc001.3gppnetwork.org.  1D IN A   10.82.10.56
> mnc001.mcc001.3gppnetwork.org.  1D IN NAPTR 10 50 "s"
> "SIP+D2U"""_sip._udp
> mnc001.mcc001.3gppnetwork.org.  1D IN NAPTR 20 50 "s"
> "SIP+D2T"""_sip._tcp
>
>
> On Mon, Jan 28, 2019 at 1:37 PM Tsvetan Filev <
> tsvetan.fi...@inno-networks.com> wrote:
>
>> Here is my current zone file:
>>
>> $ORIGIN mnc001.mcc001.3gppnetwork.org.
>> $TTL 1W
>> @   1D IN SOA   localhost. root.localhost. (
>> 1   ; serial
>> 3H  ; refresh
>> 15M ; retry
>> 1W  ; expiry
>> 1D ); minimum
>>
>> 1D IN NSns
>> ns  1D IN A 10.82.10.56
>>
>> pcscf   1D IN A 10.82.10.56
>> _sip._udp.pcscf 1D SRV 0 0 5060 pcscf
>> _sip._tcp.pcscf 1D SRV 0 0 5060 pcscf
>>
>> icscf   1D IN A 10.82.10.56
>> _sip._udp   1D SRV 0 0 4060 icscf
>> _sip._tcp   1D SRV 0 0 4060 icscf
>> _sip._udp.ims   1D SRV 0 0 4060 icscf
>> _sip._tcp.ims   1D SRV 0 0 4060 icscf
>>
>> scscf   1D IN A 10.82.10.56
>> _sip._udp.scscf 1D SRV 0 0 6060 scscf
>> _sip._tcp.scscf 1D SRV 0 0 6060 scscf
>>
>> as  1D IN A 10.82.10.56
>> _sip._udp.as1D SRV 0 0 5062 as
>> _sip._tcp.as1D SRV 0 0 5062 as
>>
>> hss 1D IN A 10.82.10.56
>>
>>
>> How do I modify it in order to make this work ?
>>
>> Tnx.
>>
>> On 28.01.19 г. 11:50 ч., Mojtaba wrote:
>>
>> Hi Tsvetan,
>> Why do you send call back to S-CSCF? You should send call back to
>> I-CSCF.  Actually in resolve of domain "mnc001.mcc001.3gppnetwork.org"
>> , The ICSCF's IP
>> should be returned.
>>  Make sure entry SRV recordd in DNS server are true.
>> This kind of call back to IMS is true, But make sure you won't have any
>> issue in DNS resolve.
>>   exten => 97255102,1,Dial(
>> SIP/972551000...@mnc001.mcc001.3gppnetwork.org,20);
>>
>> With Regards.Mojtaba
>> On Mon, Jan 28, 2019 at 12:34 PM Tsvetan Filev <
>> tsvetan.fi...@inno-networks.com> wrote:
>>
>>> Hi Mojtaba.
>>>
>>> I implemented the AS way and was able to play sound to the caller but In
>>> order to continue the call and send the invite to SCSCF I need to use proxy
>>> in the Dial application which is a problem (Asterisk is B2BUA not a proxy).
>>> I found this old question here
>>> https://community.asterisk.org/t/how-can-i-configure-asterisk-to-act-like-a-sip-proxy-server/18464
>>> that describes exactly the same issue.
>>> Here is my dial plan:
>>>
>>> exten => 97255102,1,Progress()
>>> exten => 97255102,n,Playback(vm-starmain, noanswer)
>>> exten => 97255102,n,Wait(3)
>>> exten => 97255102,n,Hangup()
>>>   ; This will send the call to the pcscf again
>>>   ;  exten => 97255102,1,Dial(
>>> SIP/972551000...@mnc001.mcc001.3gppnetwork.org,20);
>>>   ; This will send the call to scscf but it will be rejected as domain
>>> not supported
>>>   ;  exten => 97255102,1,Dial(
>>> SIP/972551000...@scscf.mnc001.mcc001.3gppnetwork.org,20);
>>>
>>> Can I use kamailio as an AS and implement the same ?
>>>
>>> Regards.
>>> On 22.12.18 г. 0:06 ч., Mojtaba wrote:
>>>
>>> Hello Tsvetan.
>>> Actually you could use SIP Early media in AS and also with cscf.
>>> If you choice the first way, i think it is very simple and
>>> strightforward because you just use early media functions on your AS. For
>>> example in Astrisk you could use Progress application and 'm' option in
>>> Dial application in your dialplan.
>

Re: [SR-Users] IMS early media

2019-01-28 Thread Mojtaba
Hi Tsvetan,
Why do you send call back to S-CSCF? You should send call back to I-CSCF.
Actually in resolve of domain "mnc001.mcc001.3gppnetwork.org"
, The ICSCF's IP should
be returned.
 Make sure entry SRV recordd in DNS server are true.
This kind of call back to IMS is true, But make sure you won't have any
issue in DNS resolve.
  exten => 97255102,1,Dial(
SIP/972551000...@mnc001.mcc001.3gppnetwork.org,20);

With Regards.Mojtaba
On Mon, Jan 28, 2019 at 12:34 PM Tsvetan Filev <
tsvetan.fi...@inno-networks.com> wrote:

> Hi Mojtaba.
>
> I implemented the AS way and was able to play sound to the caller but In
> order to continue the call and send the invite to SCSCF I need to use proxy
> in the Dial application which is a problem (Asterisk is B2BUA not a proxy).
> I found this old question here
> https://community.asterisk.org/t/how-can-i-configure-asterisk-to-act-like-a-sip-proxy-server/18464
> that describes exactly the same issue.
> Here is my dial plan:
>
> exten => 97255102,1,Progress()
> exten => 97255102,n,Playback(vm-starmain, noanswer)
> exten => 97255102,n,Wait(3)
> exten => 97255102,n,Hangup()
>   ; This will send the call to the pcscf again
>   ;  exten => 97255102,1,Dial(
> SIP/972551000...@mnc001.mcc001.3gppnetwork.org,20);
>   ; This will send the call to scscf but it will be rejected as domain not
> supported
>   ;  exten => 97255102,1,Dial(
> SIP/972551000...@scscf.mnc001.mcc001.3gppnetwork.org,20);
>
> Can I use kamailio as an AS and implement the same ?
>
> Regards.
> On 22.12.18 г. 0:06 ч., Mojtaba wrote:
>
> Hello Tsvetan.
> Actually you could use SIP Early media in AS and also with cscf.
> If you choice the first way, i think it is very simple and strightforward
> because you just use early media functions on your AS. For example in
> Astrisk you could use Progress application and 'm' option in Dial
> application in your dialplan.
> In second way you should check in Reply-Route block,if you got 180
> ringing,  you have to use rtpproxy-stream funtion for doing sip early.
>
> Wih Regards.Mojtaba Esfandiari.S
>
> On Fri, 21 Dec 2018, 16:34 Tsvetan Filev, 
> wrote:
>
>> Hi all.
>>
>> I want to use SIP early media to play music to the caller in kamailio
>> IMS installation like this:
>> http://www.sharetechnote.com/html/IMS_SIP_EarlyMedia.html
>>
>> I looked a little bit but didn't find ready solution. The information is
>> vague on this topic.
>> Should this be done through a module or application server ?
>> May I need to handle ringing in onreply_route and send OK with SDP to
>> the caller in SCSCF ?
>>
>> Regards.
>>
>>
>> ___
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>> sr-users@lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
> ___
> Kamailio (SER) - Users Mailing 
> Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>

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Re: [SR-Users] How to Register sip trunk in kamailio without username and password?

2019-04-07 Thread Mojtaba
Hello,
Kamailio doesn't work like this way. In this occasion kamailio could work
with upstream server as IP-Trust mode.
With Regards.Mojtaba

On Sun, Apr 7, 2019 at 12:42 PM Mostafa Moradi 
wrote:

> Hi, I'm mostafa moradi from iran, I Have installed kamailio, and I have a
> sip trunk without username and password, I want to register the sip trunk
> in kamailio with UAC Module, but this module only accepts sip trunk with
> user and password, How Can I Register sip trunk in kamailio without
> username and password?
>
> Thank you in advance
>
> Best regards,
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>


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Re: [SR-Users] CRITICAL ERROR in data_lamp.c!

2019-06-13 Thread Mojtaba
Great, thanks!

On Thu, 13 Jun 2019, 11:47 Daniel-Constantin Mierla  Ok, good it was sorted out.
>
> Cheers,
> Daniel
>
> On 12.06.19 12:51, Mojtaba wrote:
> > I solved it,too.
> > All issue was from didn't allocate pkg memory.
> > Thank you.
> >
> > On Wed, Jun 12, 2019 at 11:04 AM Daniel-Constantin Mierla
> >  wrote:
> >> What crash with htable api? I can't see anything related in this
> discussion.
> >>
> >> On 11.06.19 17:15, Mojtaba wrote:
> >>
> >> Thank you.
> >> And what about the crash when using it with htable api module?
> >>
> >> On Tue, 11 Jun 2019, 18:18 Daniel-Constantin Mierla  wrote:
> >>> Yes, you must allocate with pkg, because at the end those chunks are
> freed, once the new message is built. You can also look at how other
> operations with data lumps are done inside the textops module.
> >>>
> >>> On 11.06.19 14:44, Mojtaba wrote:
> >>>
> >>> I assigned a const sdp string to it just for checking. Does it matter
> to allocate new memory for newbody?
> >>>
> >>> On Tue, 11 Jun 2019, 15:22 Daniel-Constantin Mierla  wrote:
> >>>> Hello,
> >>>>
> >>>> how do you set newbody.s and newbody.len?
> >>>>
> >>>> Daniel
> >>>>
> >>>> On 11.06.19 12:29, Mojtaba wrote:
> >>>>> Not yet, because it is under developing, i could paste it here right
> away.
> >>>>>
> >>>>> {
> >>>>>  .
> >>>>>  .
> >>>>>  .
> >>>>> body.s = get_body(msg);
> >>>>> if (body.s == 0) {
> >>>>> LM_ERR("failed to get the message body\n");
> >>>>> return -1;
> >>>>> }
> >>>>> body.len = msg->len - (int) (body.s - msg->buf);
> >>>>> //body.len = strlen(body.s);
> >>>>> if (body.len == 0) {
> >>>>> LM_DBG("message body has zero length\n");
> >>>>> return -1;
> >>>>> }
> >>>>> l = del_lump(msg, body.s - msg->buf, body.len, 0);
> >>>>> if (!l) {
> >>>>> LM_ERR("del_lump failed\n");
> >>>>> return -1;
> >>>>> }
> >>>>>
> >>>>> if (!insert_new_lump_after(l, newbody.s, newbody.len, 0)) {
> >>>>> LM_ERR("insert_new_lump_after failed\n");
> >>>>> return -1;
> >>>>> }
> >>>>> }
> >>>>> I asked related issue in this regards in Kamailio developer forum,
> >>>>> before. let me paste it at below. The means of other_process function
> >>>>> is above code.
> >>>>> In during developing new module in Kamailio, I encountered with
> >>>>> strange behaviour. I used htable module instead of linked list for
> >>>>> save some temporary data during call process.dependencies module to
> >>>>> save some information. the skeleton of my code is like thus:
> >>>>>
> >>>>> if (SIP_REQUEST _INVITE) {
> >>>>>   insert_data_htable("some data");
> >>>>>   other_process(msg);
> >>>>> }else if(SIP_REPLY){
> >>>>>  get_data_htable("some data");
> >>>>>  other_process(msg);
> >>>>> }
> >>>>> In other_process function i have some changes in msg's body with
> message lump.
> >>>>> But after i make first call in kamailio. The kamailio was
> crashed.What
> >>>>> is it's problem?
> >>>>> In additional if i comment either insert/get_data_htable function or
> >>>>> othet_process function, i dont have this issue.
> >>>>> With best regards.
> >>>>>
> >>>>>
> >>>>> On Tue, Jun 11, 2019 at 2:46 PM Daniel-Constantin Mierla
> >>>>>  wrote:
> >>>>>> Hello,
> >>>>>>
> >>>>>> likely you didn't allocate the pointer inserted in data lump list.
> Is
> >>>>>> the code using data lumps available to look at on git?
> >>>>>

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