Re: [SR-Users] IPSec problem with P-CSCF

2022-10-11 Thread Mojtaba
Please check it with other IMS clients such as Boghe IMS client. It
supports all type of alg mode.
cheers

On Sun, Oct 9, 2022 at 8:58 PM Ali Akbar Hemmati 
wrote:

>  No, I'm not sure. Is there any way to check it?
>
> On Sunday, October 9, 2022, Mojtaba  wrote:
>
>> Are you sure, your client " User-Agent: IM-client/OMA1.0 HW-Rto/V1.0 "
>> use the AKAv1 algorithm for Authentication?
>>
>>
>> On Sun, Oct 9, 2022 at 12:22 PM Ali Akbar Hemmati <
>> hemmati.al...@gmail.com> wrote:
>>
>>> Yes, the pcap file is attached. The ip of the P-CSCF is 192.168.11.135.
>>> Also 192.168.11.60 is the ip from which UE reaches P-CSCF.
>>>
>>>
>>> On Sun, Oct 9, 2022 at 10:49 AM Mojtaba  wrote:
>>>
>>>> Hello Ali,
>>>> There are some critical points that should be checked for doing this
>>>> process, But would you please attach the pcap file of the registration
>>>> process,
>>>> Cheers
>>>>
>>>>
>>>> On Sat, Oct 8, 2022 at 2:24 PM Ali Akbar Hemmati <
>>>> hemmati.al...@gmail.com> wrote:
>>>>
>>>>> Hi everyone,
>>>>> I am using Kamailio as an IMS server. The UE first sends its SIP
>>>>> REGISTER over TCP and gets a 401 in response. Then the UE starts sending
>>>>> its packets over ESP. These ESP packets reach P-CSCF but are not delivered
>>>>> to its process by the kernel. We can see ESP packets in P-CSCF using
>>>>> wireshark. The ports used are correct (both client and server) based on
>>>>> their prior negotiations. We have also checked xfrm policies using the
>>>>> command `ip xfrm policy list` and seems right regarding prior 
>>>>> negotiations.
>>>>> Our Linux kernel version is 5.4 and the OS is Ubuntu 20.04.  Has anyone 
>>>>> any
>>>>> clue of the root cause of this problem? Any ideas are appreciated.
>>>>>
>>>>> Thanks,
>>>>> Aliakbar
>>>>> __
>>>>> Kamailio - Users Mailing List - Non Commercial Discussions
>>>>>   * sr-users@lists.kamailio.org
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>>>>>
>>>>
>>>>
>>>> --
>>>> --Mojtaba Esfandiari.S
>>>> __
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>>
>>
>> --
>> --Mojtaba Esfandiari.S
>>
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Re: [SR-Users] IPSec problem with P-CSCF

2022-10-09 Thread Mojtaba
Are you sure, your client " User-Agent: IM-client/OMA1.0 HW-Rto/V1.0 " use
the AKAv1 algorithm for Authentication?


On Sun, Oct 9, 2022 at 12:22 PM Ali Akbar Hemmati 
wrote:

> Yes, the pcap file is attached. The ip of the P-CSCF is 192.168.11.135.
> Also 192.168.11.60 is the ip from which UE reaches P-CSCF.
>
>
> On Sun, Oct 9, 2022 at 10:49 AM Mojtaba  wrote:
>
>> Hello Ali,
>> There are some critical points that should be checked for doing this
>> process, But would you please attach the pcap file of the registration
>> process,
>> Cheers
>>
>>
>> On Sat, Oct 8, 2022 at 2:24 PM Ali Akbar Hemmati 
>> wrote:
>>
>>> Hi everyone,
>>> I am using Kamailio as an IMS server. The UE first sends its SIP
>>> REGISTER over TCP and gets a 401 in response. Then the UE starts sending
>>> its packets over ESP. These ESP packets reach P-CSCF but are not delivered
>>> to its process by the kernel. We can see ESP packets in P-CSCF using
>>> wireshark. The ports used are correct (both client and server) based on
>>> their prior negotiations. We have also checked xfrm policies using the
>>> command `ip xfrm policy list` and seems right regarding prior negotiations.
>>> Our Linux kernel version is 5.4 and the OS is Ubuntu 20.04.  Has anyone any
>>> clue of the root cause of this problem? Any ideas are appreciated.
>>>
>>> Thanks,
>>> Aliakbar
>>> __
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>>
>>
>> --
>> --Mojtaba Esfandiari.S
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Re: [SR-Users] IPSec problem with P-CSCF

2022-10-09 Thread Mojtaba
Hello Ali,
There are some critical points that should be checked for doing this
process, But would you please attach the pcap file of the registration
process,
Cheers


On Sat, Oct 8, 2022 at 2:24 PM Ali Akbar Hemmati 
wrote:

> Hi everyone,
> I am using Kamailio as an IMS server. The UE first sends its SIP REGISTER
> over TCP and gets a 401 in response. Then the UE starts sending its packets
> over ESP. These ESP packets reach P-CSCF but are not delivered to its
> process by the kernel. We can see ESP packets in P-CSCF using wireshark.
> The ports used are correct (both client and server) based on
> their prior negotiations. We have also checked xfrm policies using the
> command `ip xfrm policy list` and seems right regarding prior negotiations.
> Our Linux kernel version is 5.4 and the OS is Ubuntu 20.04.  Has anyone any
> clue of the root cause of this problem? Any ideas are appreciated.
>
> Thanks,
> Aliakbar
> __
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Re: [SR-Users] problem getting the rtp packet source port

2022-06-23 Thread Mojtaba
I think your customers use symmetric nat, so you have to use TURN service
for relaying RTP sessions, In this situation, the ports in SDP are ignored
and the RTP will be delivered to the ports used to send.
cheers

On Mon, Jun 13, 2022 at 11:33 AM Patrick Karton 
wrote:

> Thé port in SDP is the port from which the endpoint wants to receive. It
> is not mandatory to use the same port to send traffic.
>
> Unless you know the endpoint will use symetric rtp. There is no way to
> know in avance which port it will use to send.
>
>
>
> Le 12 juin 2022 14:40, hamed r  a écrit :
>
> hello i have problem on receive source port rtp packet
> client-ip = 5.202.105.250
> server-ip = 194.5.205.33
> client2-ip = 188.212.241.46
> In the 200 ok packet from client2 ,The value 4002 is set in the audio port
> [image: photo_2022-06-12_17-44-37.jpg]
>
> I expect rtp packets to be sent to me from source port 4002, but these
> packets are sent to me from another source , which is port 55755 in the
> image below, and I just opened this port(4002) in the firewall.
> If I turn on my firewall service, all packets will be dropped. I have to
> have this firewall
> [image: photo_2022-06-12_17-44-48.jpg]
> thanks for help me
>
>
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Re: [SR-Users] Features of Kamailio as SBC

2022-01-04 Thread Mojtaba
Configuring Sems to work with Kamailio depends entirely on your network and
the scenario you want.
If you mean how Sems are configured, you can find it in the doc folder in
the source code.
cheers

On Mon, Jan 3, 2022 at 8:24 PM mohsen khashei  wrote:

> Hi can you share hkw to config sems and kamailio to work together as sbc
> (semes for signalling and kamailio+rtpengine for media)
> Thanks.
>
> On Mon, Jan 3, 2022, 8:03 PM Denys Pozniak 
> wrote:
>
>> Hello!
>>
>> We also have a similar experience with the sems, in the end, we came to
>> the conclusion that under heavy load and passing the RTP, it behaves
>> unstable.
>> Now we use sems only for SIP signaling processing and in conjunction with
>> Kamailio and Rtpengine, and in this mode it is excellent!
>>
>> If someone needs packages for Centos 7, then you can use the repository
>> (the building is triggered for each commit into sems-server master):
>> https://copr.fedorainfracloud.org/coprs/denysp/sems/
>>
>>
>> вс, 26 дек. 2021 г. в 11:28, Mojtaba :
>>
>>> I also have written some applications which use some of its SBC
>>> capabilities.
>>> I feel so happy that the SEMS works powerfully in signaling mode. For
>>> some reason, I don't anchor RTP sessions to SEMS and use a media server in
>>> Kamailio (LRKProxy).
>>> Because of the elasticity of each item, you might find some unusual
>>> manner in Sems, they are quite a topic that has no effect on Sems's overall
>>> behavior. For these cases, you can have contributions in solving them or
>>> using Kamailio.
>>>
>>> On Sun, Dec 26, 2021 at 11:17 AM Juha Heinanen  wrote:
>>>
>>>> Alex Balashov writes:
>>>>
>>>> > I use SEMS myself and like it very much. It’s sad to see it fall away
>>>> > and I regret that I personally lack the resources to prop it
>>>> > up. Nevertheless, I think it’s a fact that it’s not a dependable input
>>>> > in a future-proof way for lack of active public development and (much)
>>>> > maintenance. I think in the most optimistic view it is merely “very
>>>> > unpolished”.
>>>>
>>>> I too feel sad that SEMS development has stalled to minimal maintenance
>>>> level.  I have written numerous applications from scratch using SEMS's
>>>> DSM scripts (call center, conference, voicemail, ivr apps, charging,
>>>> ...)  plus I'm using some of its SBC capabilities.  As long as there is
>>>> no alternative, for me it is the only way to go.
>>>>
>>>> -- Juha
>>>>
>>>> __
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>>>
>>>
>>> --
>>> --Mojtaba Esfandiari.S
>>> __
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>>
>>
>> --
>>
>> BR,
>> Denys Pozniak
>>
>>
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Re: [SR-Users] Features of Kamailio as SBC

2021-12-26 Thread Mojtaba
I also have written some applications which use some of its SBC
capabilities.
I feel so happy that the SEMS works powerfully in signaling mode. For some
reason, I don't anchor RTP sessions to SEMS and use a media server in
Kamailio (LRKProxy).
Because of the elasticity of each item, you might find some unusual manner
in Sems, they are quite a topic that has no effect on Sems's overall
behavior. For these cases, you can have contributions in solving them or
using Kamailio.

On Sun, Dec 26, 2021 at 11:17 AM Juha Heinanen  wrote:

> Alex Balashov writes:
>
> > I use SEMS myself and like it very much. It’s sad to see it fall away
> > and I regret that I personally lack the resources to prop it
> > up. Nevertheless, I think it’s a fact that it’s not a dependable input
> > in a future-proof way for lack of active public development and (much)
> > maintenance. I think in the most optimistic view it is merely “very
> > unpolished”.
>
> I too feel sad that SEMS development has stalled to minimal maintenance
> level.  I have written numerous applications from scratch using SEMS's
> DSM scripts (call center, conference, voicemail, ivr apps, charging,
> ...)  plus I'm using some of its SBC capabilities.  As long as there is
> no alternative, for me it is the only way to go.
>
> -- Juha
>
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Re: [SR-Users] Features of Kamailio as SBC

2021-12-25 Thread Mojtaba
Although the Kamilio has a good performance for working as SBC, But in
combination with SEMS, it can have the best performance as an SBC.
With Best Regards.
Mojtaba Esfandiari.S

On Sat, Dec 25, 2021, 16:42 Alex Balashov  wrote:

> At the risk of self-plugging:
>
> https://youtu.be/j-0C6eHfocI
>
> http://www.evaristesys.com/blog/kamailio-as-an-sbc-five-years-on/
>
> —
> Sent from mobile, with due apologies for brevity and errors.
>
> On Dec 25, 2021, at 4:49 AM, Nattha Vudhiprasittipol <
> natth...@creaturelab.co.th> wrote:
>
> 
> Hello
>
> I would like to know all the feature lists of Kamailio as SBC.
> Could you please send me the document? I cannot find it on the website.
>
>
> Best regards
> Nattha Vudhiprasittipol
> Product Development Engineer
>
> Creaturelab Network Solution Co.,Ltd
>
> 357 Charansanitwong Rd., Bang-Bamru,
>
> Bangphlat, Bangkok 10700, THAILAND
>
>
>
> Email: natth...@creaturelab.co.th
>
> Website: www.creaturelab.co.th
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Re: [SR-Users] Issue Bug in SEMS (transparent SBC profile)

2021-12-21 Thread Mojtaba
According to RFC 3261 (section 8.1.1.5), as @Juha Heinanen mentioned, the
Cseq value can be arbitrary, It means the CSeq value can start from any
number. But according to RFC 3261 (section 22.2), as @Henning Westerholt
 mentioned, the value of CSeq must increment for 401
(Unauthorized) or 407 (Proxy Authentication Required) responses.
Then for those who want to use the Sems project, keep in mind that the Sems
does not follow these specifications.
With Best Regards

On Mon, Dec 20, 2021 at 7:43 PM Juha Heinanen  wrote:

> Mojtaba writes:
>
> > So, I have to manage the value of CSeq by Kamailio or have some update in
> > Sems so that at least could relay the Cseq without changing.
> > What do you think?
>
> If your Kamailio is between SEMS and Asterisk, you could try to increase
> CSeq value by some constant in all requests from SEMS to Asterisk.
>
> -- Juha
>


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Re: [SR-Users] Issue Bug in SEMS (transparent SBC profile)

2021-12-20 Thread Mojtaba
In the scenario that I do, I have to use transparent mode, because the
authentication is done on Asterisk servers.
And for this issue, (using the same CSeq) the authentication process is not
successful.
So, I have to manage the value of CSeq by Kamailio or have some update in
Sems so that at least could relay the Cseq without changing.
What do you think?


On Sun, Dec 19, 2021 at 5:09 PM Juha Heinanen  wrote:

> Mojtaba writes:
>
> > Yes, I agree with you, This is an exception related to 401(Unauthorized)
> > and 407 (Proxy Authentication Required) responses.
> > In these situations, The CSeq header field value should be
> > incremented.
>
> SEMS SBC does support SIP authentication, but looks like not in
> transparent mode.  From Readme.sbc.txt:
>
> SIP authentication
> --
> The SBC can perform SIP digest authentication. To use SIP authentication,
> the
> uac_auth module needs to be loaded.
>
> SIP authentication is enabled by the following parameters, separately for
> both
> call legs:
>
> # Authentication for B leg (second/callee leg):
>enable_auth   "yes" or "no"
>auth_user authentication user
>auth_pwd  authentication password
> # Authentication for A leg (first/caller leg):
>enable_aleg_auth  "yes" or "no"
>auth_aleg_userauthentication user
>auth_aleg_pwd authentication password
>
>
> Note: The 'A' leg is always the first leg, the one from the caller. 'B'
> leg is
> the one to callee:
>  caller <--- A (first) leg ---> SEMS <--- B (second) leg ---> callee
>
> Example:
>   enable_auth=yes
>   auth_user=$H(P-Auth-B-User)
>   auth_pwd=$H(P-Auth-B-Pwd)
>   enable_aleg_auth=yes
>   auth_aleg_user=$H(P-Auth-A-User)
>   auth_aleg_pwd=$H(P-Auth-A-Pwd)
>
> Perhaps yeti-switch/sems-yeti can do it also in transparent mode.
>
> -- Juha
>


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Re: [SR-Users] Issue Bug in SEMS (transparent SBC profile)

2021-12-19 Thread Mojtaba
Yes, I agree with you, This is an exception related to 401(Unauthorized)
and 407 (Proxy Authentication Required) responses.
In these situations, The CSeq header field value should be incremented.


On Sat, Dec 18, 2021 at 7:00 PM Juha Heinanen  wrote:

> Henning Westerholt writes:
>
> > I think in this particular case (SIP authentication) this is not
> > correct  (RFC 3261, 22.2):
> >
> > "When a UAC resubmits a request with its credentials after receiving a
> >401 (Unauthorized) or 407 (Proxy Authentication Required) response,
> >it MUST increment the CSeq header field value as it would normally
> >when sending an updated request."
>
> OK then, reference to this exception should have been included in
> section 8.1.1.5.
>
> -- Juha
>


-- 
-With Best Regards

*-- Mojtaba Esfandiari.S*

*-- Ph.D. student and Research Affiliate, *

*-- Technical Manager at IP-PBX Laboratory, Ferdowsi University of Mashhad,
Iran.*
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Re: [SR-Users] Issue Bug in SEMS (transparent SBC profile)

2021-12-18 Thread Mojtaba
Hello,
Let' me describe the scenario:
->--->
The UE tries to make calls, The first INVITE message is without an
Authentication header. The Asterisk server returns 401 Unauthorized.
The UE sends again INVITE messages to the asterisk server. The second
INVITE message has an Authentication header. Because both INVITE messages
have the same CSeq, the asterisk server thinks this is a LOOP message and
sends 401 Unautirozed messages again.
In both cases, the Sems set "CSeq: 10 INVITE" header, while the second the
INVITE message is not re-invite message and the CSeq should be set
incremental.


On Sat, Dec 18, 2021 at 11:15 AM Juha Heinanen  wrote:

> Mojtaba writes:
>
> > Yes, all other headers are supported. But I don't know why in the Invite
> > message, the value of CSeq is not changed, and all invite messages have
> 10
> > INVITE. This caused the asterisk server could not pass the Authentication
> > process.
>
> Are you saying that re-INVITEs have the same CSeq value as the intial
> INVITE?
>
> In initial INVITEs CSeq can have any value:
>
>8.1.1.5 CSeq
>
>For non-REGISTER requests
>outside of a dialog, the sequence number value is arbitrary.
>
> -- Juha
>
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Re: [SR-Users] Issue Bug in SEMS (transparent SBC profile)

2021-12-17 Thread Mojtaba
@Juha Heinanen:
Yes, all other headers are supported. But I don't know why in the Invite
message, the value of CSeq is not changed, and all invite messages have 10
INVITE. This caused the asterisk server could not pass the Authentication
process.

@ Denys Pozniak:
Yes, I had quick look at it, In Section 6.3 (Using Replacement in Rule),
does not mention how we can change the CSeq header.
I just imaged the value of CSeq should be incremented automatically by Sems
during Dialog, But it didn't change and at all times it has the same value
(CSeq: 10 INVITE). I need to change it because when the calling moves to
Asterisk, it should be authenticated.
Thanks

On Fri, Dec 17, 2021 at 7:22 PM Juha Heinanen  wrote:

> Mojtaba writes:
>
> > When I run SEMS, CSeq and Contact do not change, while other headers
> > change.
>
> Are you sure that changing other headers than RURI, From, To, and
> Call-ID is supported?
>
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Re: [SR-Users] Issue Bug in SEMS (transparent SBC profile)

2021-12-17 Thread Mojtaba
Any update on this?


On Sat, Dec 11, 2021, 13:56 Mojtaba  wrote:

> Hello,
> Why in SEMS, some changes (in transparent SBC profile) in sip headers do
> not apply. For example i want to change CSeq or Contact header like this:
> # transparent SBC profile
> # defaults: transparent
> RURI=$r
> From=$f
> To=$t
> Call-ID=$ci_leg2
> CSeq=$H(CSeq)
> Contact=$H(Contact)
>
> When I run SEMS, CSeq and Contact do not change, while other headers
> change.
>
> [#7fc688d7b1c0/9379] [onLoad, SBC.cpp:189] INFO:  loading SBC call
> profiles from '/usr/local/etc/sems/etc/'
>  [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:418]
> INFO:  SBC: loaded SBC profile 'mo' - MD5: d6a2c29084043edde1357231f6866737
>  [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:423]
> INFO:  SBC:  RURI  = '$r'
>  [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:424]
> INFO:  SBC:  RURI-host = ''
>  [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:425]
> INFO:  SBC:  From = '$f'
>  [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:426]
> INFO:  SBC:  To   = '$t'
>  [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:431]
> INFO:  SBC:  Call-ID   = '$ci_leg'
>  [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:434]
> INFO:  SBC:  force outbound proxy: yes
>  This issue caused the Asterisk could not pass the Authentication process
> in "match_req_to_dialog" function because the CSeq is the same as the
> dialogs. So the Asterisk thought this message was a LOOP message.
> Has anybody encountered this issue in SEMS? How did you solve this? I know
> there are some ways for doing this with Kamailio, But i focused on issues
> in SEMS
>
> With Best Regards
>
> *-- Mojtaba Esfandiari.S*
>
> *-- PhD student and Research Affiliate, *
>
> *-- Technical Manager at IP-PBX Laboratory, Ferdowsi University of
> Mashhad, Iran.*
>
> -- Address: IP-PBX Lab., Engineering Faculty, Ferdowsi University Main
> Campus, Mashhad, Iran.
>
> -- Tel: +98-51-38763635
>
> -- Mobile: +98-915-117-6713
>
>
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[SR-Users] Issue Bug in SEMS (transparent SBC profile)

2021-12-11 Thread Mojtaba
Hello,
Why in SEMS, some changes (in transparent SBC profile) in sip headers do
not apply. For example i want to change CSeq or Contact header like this:
# transparent SBC profile
# defaults: transparent
RURI=$r
From=$f
To=$t
Call-ID=$ci_leg2
CSeq=$H(CSeq)
Contact=$H(Contact)

When I run SEMS, CSeq and Contact do not change, while other headers change.

[#7fc688d7b1c0/9379] [onLoad, SBC.cpp:189] INFO:  loading SBC call profiles
from '/usr/local/etc/sems/etc/'
 [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:418] INFO:
 SBC: loaded SBC profile 'mo' - MD5: d6a2c29084043edde1357231f6866737
 [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:423] INFO:
 SBC:  RURI  = '$r'
 [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:424] INFO:
 SBC:  RURI-host = ''
 [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:425] INFO:
 SBC:  From = '$f'
 [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:426] INFO:
 SBC:  To   = '$t'
 [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:431] INFO:
 SBC:  Call-ID   = '$ci_leg'
 [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:434] INFO:
 SBC:  force outbound proxy: yes
 This issue caused the Asterisk could not pass the Authentication process
in "match_req_to_dialog" function because the CSeq is the same as the
dialogs. So the Asterisk thought this message was a LOOP message.
Has anybody encountered this issue in SEMS? How did you solve this? I know
there are some ways for doing this with Kamailio, But i focused on issues
in SEMS

With Best Regards

*-- Mojtaba Esfandiari.S*

*-- PhD student and Research Affiliate, *

*-- Technical Manager at IP-PBX Laboratory, Ferdowsi University of Mashhad,
Iran.*

-- Address: IP-PBX Lab., Engineering Faculty, Ferdowsi University Main
Campus, Mashhad, Iran.

-- Tel: +98-51-38763635

-- Mobile: +98-915-117-6713
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Re: [SR-Users] Info: Astricon 2021 - Asterisk and LRKProxy Session

2021-11-12 Thread Mojtaba
Hello there,
The Astricon videos are now available for viewing at https://
youtube.com/playlist?list=PLighc-2vlRgQTRb0PQCfFMDHuWjoUAKg4
<https://t.co/RXq2zwhJC7?amp=1> so if you missed any, head on over and give
them a watch!

cheers


On Tue, Nov 9, 2021 at 1:10 PM Mojtaba  wrote:

> Hello there,
> The videos will be published in the future by Astricon and Cluecon.
> Anyway I could send the slides here
>
> On Mon, Nov 8, 2021 at 4:53 PM Hans-Jürgen Brand <
> hans-juergen.br...@gmx.net> wrote:
>
>> Hello Mojtaba,
>>
>> can you share a link to some more infos on that?
>>
>>
>>
>> *Gesendet:* Sonntag, 07. November 2021 um 11:34 Uhr
>> *Von:* "Mojtaba" 
>> *An:* us...@lists.kamailio.org, "busin...@lists.kamailio.org" <
>> busin...@lists.kamailio.org>, "Kamailio (SER) - Development Mailing
>> List" 
>> *Betreff:* [SR-Users] Info: Astricon 2021 - Asterisk and LRKProxy Session
>> Hello,
>> The Asterisk conference (Astricon 2021) happened last week, I had a
>> session about extending Asterisk with the LRKProxy module in Kamailio in
>> terms of RTP sessions.
>> cheers,
>> --
>> --Mojtaba Esfandiari.S
>> __ Kamailio -
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>>
>
>
> --
> --Mojtaba Esfandiari.S
>


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[SR-Users] Info: Astricon 2021 - Asterisk and LRKProxy Session

2021-11-07 Thread Mojtaba
Hello,
The Asterisk conference (Astricon 2021) happened last week, I had a session
about extending Asterisk with the LRKProxy module in Kamailio in terms of
RTP sessions.
cheers,
-- 
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Re: [SR-Users] Info: ClueCon 2021 - Sessions touching Kamailio

2021-10-27 Thread Mojtaba
Great, thanks
Also, i will have a different speech about the capability of the LRKProxy
module (Media Engine) in Kamailio *in Cluecon 2021, 12:00 PM CST on
Thursday, October 28th.*
keep your eyes open

On Tue, Oct 26, 2021 at 7:07 PM Daniel-Constantin Mierla 
wrote:

> Hello,
>
> ClueCon conference happens this week, today I will have a session about
> what Kamailio offers for changing SIP message content (starting in about
> 25 minutes) and later today Fred Posner will present about Kamailio and
> FreeSwitch. You can watch the even live, more details at:
>
>   * https://www.cluecon.com/
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla -- www.asipto.com
> www.twitter.com/miconda -- www.linkedin.com/in/miconda
> Kamailio Advanced Training - Online
> Nov 08-11, 2021 (Europe Timezone) - Nov 22-25, 2021 (America Timezone)
>   * https://www.asipto.com/sw/kamailio-advanced-training-online/
>
>
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Re: [SR-Users] problem with LRKProxy module

2021-09-16 Thread Mojtaba
Hello,
The errors show the lrkproxy module in kamailio could not connect with
lrkproxy service in udp:192.168.51.106:22333.
Make sure they are connected to each other by the PING-PONG message command
in syslog.

On Thu, Sep 16, 2021 at 12:40 PM hamed r  wrote:

> Hello
> I have problem with lrkproxy module.
> First, I installed the packages from the link below :
> https://github.com/mojtabaesfandiari/pylrkproxy
> And the service was properly listened to the local ip (
> 192.168.51.106:22333).
>
> In the next step, the parameters were added to the kamalio.cfg file :
> loadmodule "lrkproxy.so"
> modparam("lrkproxy", "lrkproxy_sock", "udp:192.168.51.106:22333")
>
> I also changed the  /etc/pylrkproxy/pylrkproxy.ini to set local ip.
>
> But when I make a call, the ip inside the sdp not change.
> And the following error is displayed in the syslog:
> Sep  4 07:18:05 debian /usr/local/sbin/kamailio[1724]: INFO: {1 2 INVITE
> NzA0NmU0YTYwNWNkNzAwOWJiOTUyOWQ5ODhjNjRhNzk.} lrkproxy [lrkproxy.c:1507]:
> lrkproxy_force(): Here is SIP_REQUEST &  METHOD_INVITE
> Sep  4 07:18:05 debian /usr/local/sbin/kamailio[1724]: ERROR: {1 2 INVITE
> NzA0NmU0YTYwNWNkNzAwOWJiOTUyOWQ5ODhjNjRhNzk.} lrkproxy [lrkproxy.c:1084]:
> select_lrkp_node(): script error -no valid set selected
> Sep  4 07:18:05 debian /usr/local/sbin/kamailio[1724]: ERROR: {1 2 INVITE
> NzA0NmU0YTYwNWNkNzAwOWJiOTUyOWQ5ODhjNjRhNzk.} lrkproxy [lrkproxy.c:1513]:
> lrkproxy_force(): can't ping any lrk proxy right now.
> Sep  4 07:18:07 debian /usr/local/sbin/kamailio[1723]: INFO: {2 2 INVITE
> NzA0NmU0YTYwNWNkNzAwOWJiOTUyOWQ5ODhjNjRhNzk.} lrkproxy [lrkproxy.c:1625]:
> lrkproxy_force(): Here is SIP_REPLY of METHOD_INVITE
> Sep  4 07:18:07 debian /usr/local/sbin/kamailio[1723]: ERROR: {2 2 INVITE
> NzA0NmU0YTYwNWNkNzAwOWJiOTUyOWQ5ODhjNjRhNzk.} lrkproxy [lrkproxy.c:1630]:
> lrkproxy_force(): No found entry in hash table
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Re: [SR-Users] How tell if rtp packets are being relayed through rtpengine in kamailio

2021-08-23 Thread Mojtaba
Hello,
The Kamailio just works  as SIPProxy and it does not relay RTP packets at
all. Some services are available as module in Kamailio for relaying RTP
packets like as rtpproxy, rtpengine, lrkproxy. In all cases, you have to
set WITH_NAT option in Kamailio configuration.
With Regards.
Mojtaba

On Mon, Aug 23, 2021 at 10:42 AM Faiz Fazaluddin 
wrote:

> Hi,
> I want to see if rtp packets are being relayed through kamailio. I am
> attmepting to connect rtpengine with kamailio, but am not sure if it is
> working, how can I tell?
>
> I have the kamailio server on a different VM than the clients. In
> wireshark, no rtp packets show on the server VM and rtp packets only seem
> to be going directly through the client VM.
>
> Rtp engine mentions being unable to find hashtables when attempting to
> implement a delete_node command.
>
> Thank you very much for your help,
> Faiz
>
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Re: [SR-Users] install latest sems-server on debian 10 buster

2021-08-18 Thread Mojtaba
Sure.

On Mon, Aug 16, 2021, 01:24 Denys Pozniak  wrote:

> @Mojtaba if not difficult, could you add a description of the installation
> process to the wiki?
> https://github.com/sems-server/sems/wiki
>
>
>
> вс, 15 авг. 2021 г. в 11:06, Mojtaba :
>
>> After several attempts to install SEMS in Debian10, it was done
>> successfully.
>> The issue was strang but I had to install "python3.7-dev
>> python3-distutils", and update PATH in Linux.
>> Thanks
>>
>> On Thu, Aug 12, 2021 at 12:32 PM Mojtaba  wrote:
>>
>>> I installed the master branch of SEMS(1.7.0~dev) as you suggested. I
>>> used these commands:
>>> $ git clone https://github.com/sems-server/sems.git
>>> $ cd sems/
>>> $  ln -s pkg/deb/buster ./debian
>>> $ dch -v 1.7.0~dev "SEMS 1.7.0~dev release"
>>> $ dpkg-buildpackage -rfakeroot -us -uc
>>>
>>> Does it need to run "make & make install" commands after above commands
>>> or not? because i got these error as i told
>>>
>>> usr/bin/ld: AmDtmfDetector.o: in function
>>> `AmSpanDSPInbandDtmfDetector::streamPut(unsigned char const*, unsigned int,
>>> unsigned long long)':
>>> ./core/AmDtmfDetector.cpp:768: undefined reference to `dtmf_rx'
>>> /usr/bin/ld: AmDtmfDetector.o: in function
>>> `AmSpanDSPInbandDtmfDetector::AmSpanDSPInbandDtmfDetector(AmKeyPressSink*,
>>> int)':
>>> ./core/AmDtmfDetector.cpp:749: undefined reference to `dtmf_rx_init'
>>> /usr/bin/ld: AmDtmfDetector.o: in function
>>> `AmSpanDSPInbandDtmfDetector::~AmSpanDSPInbandDtmfDetector()':
>>> ./core/AmDtmfDetector.cpp:763: undefined reference to `dtmf_rx_free'
>>> /usr/bin/ld: AmDtmfDetector.o: in function
>>> `AmSpanDSPInbandDtmfDetector::AmSpanDSPInbandDtmfDetector(AmKeyPressSink*,
>>> int)':
>>> ./core/AmDtmfDetector.cpp:754: undefined reference to
>>> `dtmf_rx_set_realtime_callback'
>>> collect2: error: ld returned 1 exit status
>>> make[1]: *** [Makefile:115: sems] Error 1
>>> make[1]: Leaving directory '/usr/src/sems-master/core'
>>> make: *** [Makefile:38: install] Error 2
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Tue, Aug 10, 2021 at 4:24 PM Juha Heinanen  wrote:
>>>
>>>> Mojtaba writes:
>>>>
>>>> > Would you please let me know about what version of sems you use?
>>>> > Please tell me more about it.
>>>>
>>>> Master branch of https://github.com/sems-server/sems.
>>>>
>>>> -- Juha
>>>>
>>>> __
>>>> Kamailio - Users Mailing List - Non Commercial Discussions
>>>>   * sr-users@lists.kamailio.org
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>>>> to the sender!
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>>>
>>>
>>> --
>>> --Mojtaba Esfandiari.S
>>>
>>
>>
>> --
>> --Mojtaba Esfandiari.S
>> __
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>
>
> --
>
> BR,
> Denys Pozniak
>
>
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Re: [SR-Users] install latest sems-server on debian 10 buster

2021-08-15 Thread Mojtaba
After several attempts to install SEMS in Debian10, it was done
successfully.
The issue was strang but I had to install "python3.7-dev
python3-distutils", and update PATH in Linux.
Thanks

On Thu, Aug 12, 2021 at 12:32 PM Mojtaba  wrote:

> I installed the master branch of SEMS(1.7.0~dev) as you suggested. I used
> these commands:
> $ git clone https://github.com/sems-server/sems.git
> $ cd sems/
> $  ln -s pkg/deb/buster ./debian
> $ dch -v 1.7.0~dev "SEMS 1.7.0~dev release"
> $ dpkg-buildpackage -rfakeroot -us -uc
>
> Does it need to run "make & make install" commands after above commands or
> not? because i got these error as i told
>
> usr/bin/ld: AmDtmfDetector.o: in function
> `AmSpanDSPInbandDtmfDetector::streamPut(unsigned char const*, unsigned int,
> unsigned long long)':
> ./core/AmDtmfDetector.cpp:768: undefined reference to `dtmf_rx'
> /usr/bin/ld: AmDtmfDetector.o: in function
> `AmSpanDSPInbandDtmfDetector::AmSpanDSPInbandDtmfDetector(AmKeyPressSink*,
> int)':
> ./core/AmDtmfDetector.cpp:749: undefined reference to `dtmf_rx_init'
> /usr/bin/ld: AmDtmfDetector.o: in function
> `AmSpanDSPInbandDtmfDetector::~AmSpanDSPInbandDtmfDetector()':
> ./core/AmDtmfDetector.cpp:763: undefined reference to `dtmf_rx_free'
> /usr/bin/ld: AmDtmfDetector.o: in function
> `AmSpanDSPInbandDtmfDetector::AmSpanDSPInbandDtmfDetector(AmKeyPressSink*,
> int)':
> ./core/AmDtmfDetector.cpp:754: undefined reference to
> `dtmf_rx_set_realtime_callback'
> collect2: error: ld returned 1 exit status
> make[1]: *** [Makefile:115: sems] Error 1
> make[1]: Leaving directory '/usr/src/sems-master/core'
> make: *** [Makefile:38: install] Error 2
>
>
>
>
>
>
>
>
> On Tue, Aug 10, 2021 at 4:24 PM Juha Heinanen  wrote:
>
>> Mojtaba writes:
>>
>> > Would you please let me know about what version of sems you use?
>> > Please tell me more about it.
>>
>> Master branch of https://github.com/sems-server/sems.
>>
>> -- Juha
>>
>> __
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>
>
> --
> --Mojtaba Esfandiari.S
>


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Re: [SR-Users] install latest sems-server on debian 10 buster

2021-08-12 Thread Mojtaba
I installed the master branch of SEMS(1.7.0~dev) as you suggested. I used
these commands:
$ git clone https://github.com/sems-server/sems.git
$ cd sems/
$  ln -s pkg/deb/buster ./debian
$ dch -v 1.7.0~dev "SEMS 1.7.0~dev release"
$ dpkg-buildpackage -rfakeroot -us -uc

Does it need to run "make & make install" commands after above commands or
not? because i got these error as i told

usr/bin/ld: AmDtmfDetector.o: in function
`AmSpanDSPInbandDtmfDetector::streamPut(unsigned char const*, unsigned int,
unsigned long long)':
./core/AmDtmfDetector.cpp:768: undefined reference to `dtmf_rx'
/usr/bin/ld: AmDtmfDetector.o: in function
`AmSpanDSPInbandDtmfDetector::AmSpanDSPInbandDtmfDetector(AmKeyPressSink*,
int)':
./core/AmDtmfDetector.cpp:749: undefined reference to `dtmf_rx_init'
/usr/bin/ld: AmDtmfDetector.o: in function
`AmSpanDSPInbandDtmfDetector::~AmSpanDSPInbandDtmfDetector()':
./core/AmDtmfDetector.cpp:763: undefined reference to `dtmf_rx_free'
/usr/bin/ld: AmDtmfDetector.o: in function
`AmSpanDSPInbandDtmfDetector::AmSpanDSPInbandDtmfDetector(AmKeyPressSink*,
int)':
./core/AmDtmfDetector.cpp:754: undefined reference to
`dtmf_rx_set_realtime_callback'
collect2: error: ld returned 1 exit status
make[1]: *** [Makefile:115: sems] Error 1
make[1]: Leaving directory '/usr/src/sems-master/core'
make: *** [Makefile:38: install] Error 2








On Tue, Aug 10, 2021 at 4:24 PM Juha Heinanen  wrote:

> Mojtaba writes:
>
> > Would you please let me know about what version of sems you use?
> > Please tell me more about it.
>
> Master branch of https://github.com/sems-server/sems.
>
> -- Juha
>
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Re: [SR-Users] install latest sems-server on debian 10 buster

2021-08-10 Thread Mojtaba
Would you please let me know about what version of sems you use?
Please tell me more about it.
Thanks

On Mon, Aug 9, 2021 at 10:49 AM Juha Heinanen  wrote:

> Mojtaba writes:
>
> > ./core/AmDtmfDetector.cpp:768: undefined reference to `dtmf_rx'
>
> That is defined in libspandsp.  The control file below works for me in
> Debian Buster.
>
> -- Juha
>
> Source: opensipg-sems
> Section: net
> Priority: optional
> Maintainer: Juha Heinanen 
> Build-Depends: debhelper (>= 9), libcurl4-openssl-dev |
> libcurl4-gnutls-dev, libmysqlcppconn-dev, libspandsp-dev, libspeex-dev,
> python-mysqldb, libmpg123-dev, libssl-dev, libev-dev, libmp3lame-dev,
> libevent-dev (>= 2.0.0), libopus-dev, openssl, python-dev, python-sip-dev,
> libhiredis-dev, libbcg729-dev (>= 1.1.0)
> Standards-Version: 3.5.2
>
> Package: opensipg-sems
> Architecture: any
> Depends: ${shlibs:Depends}, ${misc:Depends}, python-mysqldb, lame, speex,
> php-cli, mpg123, sox, m4, opensipg-utils
> Description: Application Server for OpenSIPg
>  Provides voicemail, announcement, conferencing, etc. applications.
>
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Re: [SR-Users] help

2021-08-08 Thread Mojtaba
Hello there,
You should do the following steps:
1- Add new AS.
2- Add some SPT as a Trigger Points.
3- Create IFC linked to AS and Trigger Points.
For more information go  to ETSI document,
With best regards
Mojtaba Esfandiari.S


On Sun, Aug 1, 2021 at 12:40 AM Karsten Horsmann 
wrote:

> Hi Said,
>
> if no one is answering your questions then you should ask more specific
> what is the issue and what you try to solve that.
>
> That could result in more answers.
>
> Kind regards
> Karsten Horsmann
>
> Said Hassani  schrieb am Fr., 30. Juli 2021,
> 08:11:
>
>> Dear Team,
>>
>> I would like to know how to link Kamailio with openIMSCore?
>> I would like to use Kamailio as an Application server.
>> Please guide me or if there is someone who has worked on this subject, I
>> would like to benefit from his help.
>>
>> __
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Re: [SR-Users] install latest sems-server on debian 10 buster

2021-08-08 Thread Mojtaba
Although during installing *sems 1.7.0~dev, *some Errors were happened in
AmDtmfDetector.cpp file, But by continuing the installing dpkg files in ../
path, SEMS is installed and worked
libsems1-dev_1.7.0~dev_amd64.deb
sems_1.7.0~dev_amd64.deb
sems-dbg_1.7.0~dev_amd64.deb

Is this the right way to install?
Thanks


On Sat, Aug 7, 2021 at 12:07 PM Mojtaba  wrote:

> Sure,
> *1) The issue related to  SEMS 1.7.0~dev in Linux debian10 4.19.0-6-amd64:*
> *I run these command in order:*
> *$ ln -s pkg/deb/buster ./debian*
> *$ dch -v 1.7.0~dev "SEMS 1.7.0~dev release"*
> *$ dpkg-buildpackage -rfakeroot -us -uc*
> *And then try to use make and make install. I got these Errors:*
>
> making sems executable
> gcc -o sems AmB2BSession.o AmAppTimer.o AmSessionProcessor.o AmEvent.o
> AmAudio.o AmUACAuth.o AmB2BMedia.o AmB2ABSession.o AmAudioMixer.o
> AmBufferedAudio.o AmConferenceChannel.o AmEventDispatcher.o
> AmPromptCollection.o SipCtrlInterface.o AmEventQueueProcessor.o md5.o
> crc4.o AmAudioMixIn.o AmRtpMuxStream.o AmEventProcessingThread.o AmArg.o
> AmZRTP.o AmCachedAudioFile.o AmMediaProcessor.o AmSession.o
> AmSessionEventHandler.o AmOfferAnswer.o AmConfigReader.o
> AmSipSubscriptionContainer.o AmThread.o AmEventQueue.o jsonxx.o
> AmSipSubscription.o AmSipDialog.o AmSipEvent.o AmApi.o AmRtpPacket.o
> AmUriParser.o AmSdp.o Am100rel.o AmRingTone.o AmDtmfSender.o AmRtpStream.o
> AmSipDispatcher.o AmPeriodicThread.o AmUAC.o AmMimeBody.o
> AmSipRegistration.o sems.o AmRtpAudio.o log.o AmPlaylist.o
> AmBasicSipDialog.o AmPrecodedFile.o AmSessionContainer.o AmDtmfDetector.o
> AmPlayoutBuffer.o AmPlugIn.o AmUtils.o jsonArg.o AmSipMsg.o
> AmMultiPartyMixer.o AmAudioFile.o AmConfig.o AmCallWatcher.o
> AmAdvancedAudio.o AmJitterBuffer.o AmRtpReceiver.o AmConferenceStatus.o
> LowcFE.o resample/libresample.a -lm -ldl -rdynamic -lpthread -lstdc++
>  -levent -levent_pthreads -Wl,--whole-archive sip/sip_stack.a
> -Wl,--no-whole-archive -lresolv
> /usr/bin/ld: AmDtmfDetector.o: in function
> `AmSpanDSPInbandDtmfDetector::streamPut(unsigned char const*, unsigned int,
> unsigned long long)':
> ./core/AmDtmfDetector.cpp:768: undefined reference to `dtmf_rx'
> /usr/bin/ld: AmDtmfDetector.o: in function
> `AmSpanDSPInbandDtmfDetector::AmSpanDSPInbandDtmfDetector(AmKeyPressSink*,
> int)':
> ./core/AmDtmfDetector.cpp:749: undefined reference to `dtmf_rx_init'
> /usr/bin/ld: AmDtmfDetector.o: in function
> `AmSpanDSPInbandDtmfDetector::~AmSpanDSPInbandDtmfDetector()':
> ./core/AmDtmfDetector.cpp:763: undefined reference to `dtmf_rx_free'
> /usr/bin/ld: AmDtmfDetector.o: in function
> `AmSpanDSPInbandDtmfDetector::AmSpanDSPInbandDtmfDetector(AmKeyPressSink*,
> int)':
> ./core/AmDtmfDetector.cpp:754: undefined reference to
> `dtmf_rx_set_realtime_callback'
> collect2: error: ld returned 1 exit status
> make[1]: *** [Makefile:115: sems] Error 1
> make[1]: Leaving directory '/usr/src/sems/core'
> make: *** [Makefile:27: modules] Error 2
>
> *2) The issue related to latest SEMS 1.6.0 in Linux debian10
> 4.19.0-6-amd64:*
> *$ dpkg-checkbuilddeps*
> *dpkg-checkbuilddeps: error: Unmet build dependencies: libmysql++-dev*
> Because of the library of libmysql++-dev is renamed to libmysqlcppconn-dev
> in debian 10, i updated it's name in control file like below:
> Build-Depends: debhelper (>= 9~),
>flite-dev,
>libcurl4-openssl-dev | libcurl4-gnutls-dev,
>libev-dev,
>libevent-dev (>= 2.0.0),
>libhiredis-dev,
> *   libmysqlcppconn-dev,*
>libspandsp-dev,
>libspeex-dev,
>libssl-dev,
>libxml2-dev,
>openssl,
>python-dev,
>python-sip-dev
> Standards-Version: 3.9.5
>
> Then i run these commands:
> $dch -v 1.6.0 "SEMS 1.6.0 release"
> $ dpkg-buildpackage -rfakeroot -us -uc
> This error is appeared:
> making sems executable
> gcc -o sems LowcFE.o AmB2BSession.o AmAppTimer.o AmSessionProcessor.o
> AmEvent.o AmAudio.o AmUACAuth.o AmB2BMedia.o AmB2ABSession.o AmAudioMixer.o
> AmBufferedAudio.o AmConferenceChannel.o AmEventDispatcher.o
> AmPromptCollection.o SipCtrlInterface.o AmEventQueueProcessor.o md5.o
> AmAudioMixIn.o AmEventProcessingThread.o AmArg.o AmZRTP.o
> AmCachedAudioFile.o AmMediaProcessor.o AmSession.o AmSessionEventHandler.o
> AmOfferAnswer.o sems.o AmConfigReader.o AmSipSubscriptionContainer.o
> AmThread.o AmEventQueue.o jsonxx.o AmSipSubscription.o AmSipDialog.o
> AmSipEvent.o AmUriParser.o AmSdp.o AmRingTone.o AmDtmfSender.o
> AmRtpStream.o AmSipDispatcher.o AmPeriodicThread.o AmUAC.o Am100rel.o
> AmMimeBody.o AmSipRegistration.o AmRtpAudio.o log.o A

Re: [SR-Users] install latest sems-server on debian 10 buster

2021-08-07 Thread Mojtaba
ge: error: fakeroot debian/rules binary subprocess returned
exit status 2

Let me know if you need more issue logs.
Thank you



On Sat, Aug 7, 2021 at 1:21 AM Antony Stone <
antony.st...@kamailio.open.source.it> wrote:

> On Friday 06 August 2021 at 22:43:59, Mojtaba wrote:
>
> > Does anybody have experience installing the latest SEMS-Server (1.6.0) on
> > debian 10 buster?
> > I tried to install both 1.6.0. and 1.7-dev and some issues are occurred,
>
> Hm, maybe tell us what "issues" means?
>
>
> Antony.
>
> --
> Angela Merkel arrives at Paris airport.
> "Nationality?" asks the immigration officer.
> "German," she replies.
> "Occupation?"
> "No, just here for a summit conference."
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
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[SR-Users] install latest sems-server on debian 10 buster

2021-08-06 Thread Mojtaba
Hello there,
Does anybody have experience installing the latest SEMS-Server (1.6.0) on
debian 10 buster?
I tried to install both 1.6.0. and 1.7-dev and some issues are occurred,
but i installed the version 1.3.1 on debian 8 before.
Any help would be appreciated.
Thanks with best regards

-- 
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Re: [SR-Users] which is better for B2BUA

2021-08-02 Thread Mojtaba
Hi Carsten,
Yes, Using SEMS as B2BUA would be good plan, But just a question, Do you
skip RTP flow to SEMS, too?
Thanks

On Mon, Aug 2, 2021 at 1:46 PM Carsten Bock  wrote:

> Hi,
>
> we even take it one step further:
> Incoming <-> Kamailio <-> SEMS (B2BUA) <-> Kamailio <-> Interconnect
>
> We do this, to limit the required functionality of SEMS to be a B2BUA and
> we do all further stuff (Number Portability Lookup, Least-Cost-Routing,
> Message Manipulation, ..) on Kamailio.
>
> I almost hate to say this as Kamailian, but one more alternative to SEMS
> worth checking is OpenSIPS and its b2b modules.
>
> Thanks,
> Carsten
> --
> Carsten Bock I CTO & Founder
>
> ng-voice GmbH
>
> Trostbrücke 1 I 20457 Hamburg I Germany
> T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com
>
> Registry Office at Local Court Hamburg, HRB 120189
> Managing Directors: Dr. David Bachmann, Carsten Bock
>
>
> Am Sa., 31. Juli 2021 um 23:24 Uhr schrieb Mojtaba :
>
>> Thanks Guys for your comments.
>> @Gerry, I agree with you, I have a few experiences working with Yate, It
>> could work great in enterprise solutions. Thanks
>> @Denys, Thanks for sharing your experience, Although its configurations
>> look complicated, But in signaling, It would work great.
>> Just a question, Let me know which scenario is like that you use?
>>
>> ---
>> or
>> ---
>>
>>
>>
>> On Fri, Jul 30, 2021 at 9:15 AM Denys Pozniak 
>> wrote:
>>
>>> Hello!
>>> We used sems to interconnect with external operators. But with a heavy
>>> call load and with the processing of the media stream, it often crashed.
>>> Now we use Kamailio / Rtpengine for media stream processing and sems for
>>> signaling processing. We install all this on one server. It probably looks
>>> complicated, but it works great.
>>>
>>> вт, 27 июл. 2021 г. в 16:10, Gerry | Rigatta :
>>>
>>>> Hi,
>>>>
>>>> Some comments on Yate.
>>>>
>>>> Yate works well as a B2BUA. It is very performant and stable. It can
>>>> bypass media dynamically.
>>>>
>>>> The main beauty of Yate is how it processes calls internally. Yate
>>>> represents calls internally as messages with parameters (caller, callee,
>>>> codecs …) . While the call passes from the incoming to the outgoing call
>>>> leg the message parameters can be manipulated by different yate modules. In
>>>> turn, message parameters can trigger yate modules for some actions, e.g.
>>>> start a sip call.
>>>>
>>>> The Yate internal messaging allows to build with little code complex
>>>> call logic. E.g a mysql query result can be used directly for routing. E,g,
>>>> you can build a switch just with a couple of mysql procedures.
>>>>
>>>> Cheers
>>>>
>>>> Gerry
>>>>
>>>>
>>>>
>>>> On 27 Jul 2021, at 11:35, Karsten Horsmann  wrote:
>>>>
>>>> Hello,
>>>>
>>>> just for the records:
>>>>
>>>> You can also bypass media with FreeSWITCH if you want. The overall
>>>> handling of many parallel calls is still not so high like in SEMS.
>>>>
>>>> https://freeswitch.org/confluence/display/FREESWITCH/Proxy+Media
>>>> This is also possible on a per Call base.
>>>>
>>>> And you can use different profiles as well in FreeSWITCH.
>>>>
>>>> Asterisk Experts will have more knowledge what is possible there, and
>>>> maybe someone told us whats going on with Yate.
>>>>
>>>> Kind regards
>>>> Karsten
>>>>
>>>> Am Di., 13. Juli 2021 um 10:11 Uhr schrieb Mojtaba :
>>>>
>>>>>
>>>>> Hello there,
>>>>> According to ,
>>>>> https://lists.kamailio.org/pipermail/sr-users/2016-March/092058.html,
>>>>> which talked about  B2BUA (just signalling) in Kamailio.
>>>>> As i have experienced working with SEMS, freeswitch and Kamailio while
>>>>> using B2BUA feature, Each of them have pros and cons:
>>>>> 1- The sems is a light sip engine server with several applications
>>>>> (like as sbc) for using b2bua. All incoming and outgoing calls could go to
>>>>> sems server for doing b2bua like this:
>>>>>
>>>>>
>>

Re: [SR-Users] which is better for B2BUA

2021-07-31 Thread Mojtaba
Thanks Guys for your comments.
@Gerry, I agree with you, I have a few experiences working with Yate, It
could work great in enterprise solutions. Thanks
@Denys, Thanks for sharing your experience, Although its configurations
look complicated, But in signaling, It would work great.
Just a question, Let me know which scenario is like that you use?

---
or
---



On Fri, Jul 30, 2021 at 9:15 AM Denys Pozniak 
wrote:

> Hello!
> We used sems to interconnect with external operators. But with a heavy
> call load and with the processing of the media stream, it often crashed.
> Now we use Kamailio / Rtpengine for media stream processing and sems for
> signaling processing. We install all this on one server. It probably looks
> complicated, but it works great.
>
> вт, 27 июл. 2021 г. в 16:10, Gerry | Rigatta :
>
>> Hi,
>>
>> Some comments on Yate.
>>
>> Yate works well as a B2BUA. It is very performant and stable. It can
>> bypass media dynamically.
>>
>> The main beauty of Yate is how it processes calls internally. Yate
>> represents calls internally as messages with parameters (caller, callee,
>> codecs …) . While the call passes from the incoming to the outgoing call
>> leg the message parameters can be manipulated by different yate modules. In
>> turn, message parameters can trigger yate modules for some actions, e.g.
>> start a sip call.
>>
>> The Yate internal messaging allows to build with little code complex call
>> logic. E.g a mysql query result can be used directly for routing. E,g, you
>> can build a switch just with a couple of mysql procedures.
>>
>> Cheers
>>
>> Gerry
>>
>>
>>
>> On 27 Jul 2021, at 11:35, Karsten Horsmann  wrote:
>>
>> Hello,
>>
>> just for the records:
>>
>> You can also bypass media with FreeSWITCH if you want. The overall
>> handling of many parallel calls is still not so high like in SEMS.
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/Proxy+Media
>> This is also possible on a per Call base.
>>
>> And you can use different profiles as well in FreeSWITCH.
>>
>> Asterisk Experts will have more knowledge what is possible there, and
>> maybe someone told us whats going on with Yate.
>>
>> Kind regards
>> Karsten
>>
>> Am Di., 13. Juli 2021 um 10:11 Uhr schrieb Mojtaba :
>>
>>>
>>> Hello there,
>>> According to ,
>>> https://lists.kamailio.org/pipermail/sr-users/2016-March/092058.html,
>>> which talked about  B2BUA (just signalling) in Kamailio.
>>> As i have experienced working with SEMS, freeswitch and Kamailio while
>>> using B2BUA feature, Each of them have pros and cons:
>>> 1- The sems is a light sip engine server with several applications (like
>>> as sbc) for using b2bua. All incoming and outgoing calls could go to sems
>>> server for doing b2bua like this:
>>>
>>>
>>> Incoming<===>Kamailio<>Sems<>Kamailio<===>outgoing
>>>
>>> 2- In sems, you could disable rtp realying. It forces sems to work just
>>> as b2bua without anchoring RTP
>>> 3- Easy to use different active profiles in routing.
>>>
>>> __
>> Kamailio - Users Mailing List - Non Commercial Discussions
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>>
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>
>
> --
>
> BR,
> Denys Pozniak
>
>
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Re: [SR-Users] Active Profile for sipreply in SEMS

2021-07-17 Thread Mojtaba
Of course, It could be added in different profiles based on different
active_profile methods, But it does not work for 200 ok reply messages. In
those cases, i added SBC to whitelist header filter, like this:
header_filter=whitelist
header_list=Diversion,P-Asserted-Identity,Privacy,Allow,Event,Expires,Accept,User-Agent,Subscription-State,P-Access-Network-Info,P-Route,C-Params,Feature-Caps,SBC
Actually I solved it by changing the header name to P-Route, because the
P_Route header is in the whitelist header by default in sipreply messages.
But what about custom header name? What should i do?
Thanks With Regards.
Mojtaba


On Sat, Jul 17, 2021 at 3:21 PM Arsen Semenov  wrote:

> Hi Mojatba,
>
> basically you should be able to set whitelist in your sbc profile like
> this:
>
> header_filter=whitelist
> header_list=SBC
>
> Not sure how it will work for replies however..
> you also can try "transparent" - no filtering mode
>
>
> Regards,
>
>
> On Sat, Jul 17, 2021 at 2:50 PM Mojtaba  wrote:
>
>> Hello,
>> Does anybody know how i could change white filter in siprelpy in sems?
>> In Kamailio, I got "200 OK" from UE, then add a new header (SBC:
>> mt-reply) and send it to sems server. The sems server remove it:
>>  [#7f410fbfb700/4560] [inplaceHeaderFilter, HeaderFilter.cpp:226] DEBUG:
>>  erasing header 'sbc' by whitelist
>>
>> I really don't want to remove it.
>> Actually if I knew how i active profile for sip replying in sems, I would
>> be able to solve this.
>>
>> Thanks with regards.
>> --
>> --Mojtaba Esfandiari.S
>> __
>> Kamailio - Users Mailing List - Non Commercial Discussions
>>   * sr-users@lists.kamailio.org
>> Important: keep the mailing list in the recipients, do not reply only to
>> the sender!
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>
>
> --
> Arsen Semenov
>
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[SR-Users] Active Profile for sipreply in SEMS

2021-07-17 Thread Mojtaba
Hello,
Does anybody know how i could change white filter in siprelpy in sems?
In Kamailio, I got "200 OK" from UE, then add a new header (SBC: mt-reply)
and send it to sems server. The sems server remove it:
 [#7f410fbfb700/4560] [inplaceHeaderFilter, HeaderFilter.cpp:226] DEBUG:
 erasing header 'sbc' by whitelist

I really don't want to remove it.
Actually if I knew how i active profile for sip replying in sems, I would
be able to solve this.

Thanks with regards.
-- 
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[SR-Users] which is better for B2BUA

2021-07-13 Thread Mojtaba
Hello there,
According to ,
https://lists.kamailio.org/pipermail/sr-users/2016-March/092058.html, which
talked about  B2BUA (just signalling) in Kamailio.
As i have experienced working with SEMS, freeswitch and Kamailio while
using B2BUA feature, Each of them have pros and cons:
1- The sems is a light sip engine server with several applications (like as
sbc) for using b2bua. All incoming and outgoing calls could go to sems
server for doing b2bua like this:

Incoming<===>Kamailio<>Sems<>Kamailio<===>outgoing

2- In sems, you could disable rtp realying. It forces sems to work just as
b2bua without anchoring RTP
3- Easy to use different active profiles in routing.
Just a couple of things there are in SEMS.  For example, the sems adds
itself (local IP) in Via header, and it couldn't be common in b2bua. like
this:
.
.
.
INVITE sip:2...@cloud.domain.com;transport=UDP SIP/2.0
Via: SIP/2.0/UDP
10.10.10.200;branch=z9hG4bK2f53.07a3fd9edaa8c8d609ab2ac6b01a087f.0
*>Kamailio
public IP*
*Via: SIP/2.0/UDP
127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKsPRMNast;rport=5080
===>private loopback sems ip*
From: ;tag=6FFDB493-60EABB3600016ECD-379F9700
To: 
CSeq: 10 INVITE
Call-ID: Y2M1ODQxNTZmMjdkZWZjN2U5MmMyYjBmN2Y2OGY1ODQ._leg2
Route: 
Max-Forwards: 69
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
User-Agent: Z 3.3.25608 r25552
Content-Type: application/sdp
Contact: 
Content-Length: 246
.
.
.
For this reason, Is there a way to avoid this issue? I know it is possible
to do this by using other modules like textops and retransformation ?
And why in Kamailio, there is no b2bua module to perform all b2bua
functionality, yet?
Thanks with Best Regards.


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Re: [SR-Users] Sipwise SEMS

2021-07-12 Thread Mojtaba
Hello there,
As i have experienced working with SEMS, freeswitch and Kamailio while
using B2BUA feature, Each of them have pros and cons:
1- The sems is a light sip engine server with several applications (like as
sbc) for using b2bua. All incoming and outgoing calls could go to sems
server for doing b2bua like this:

Incoming<===>Kamailio<>Sems<>Kamailio<===>outgoing

2- In sems, you could disable rtp realying. It forces sems to work just as
b2bua without anchoring RTP
3- Easy to use different active profiles in routing.

Just a couple of things there are in SEMS.  For example, the sems adds
itself (local IP) in Via header, and it couldn't be common in b2bua. like
this:
.
.
.
INVITE sip:2...@cloud.domain.com;transport=UDP SIP/2.0
Via: SIP/2.0/UDP
10.10.10.200;branch=z9hG4bK2f53.07a3fd9edaa8c8d609ab2ac6b01a087f.0
*>Kamailio
public IP*
*Via: SIP/2.0/UDP
127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKsPRMNast;rport=5080
===>private loopback sems ip*
From: ;tag=6FFDB493-60EABB3600016ECD-379F9700
To: 
CSeq: 10 INVITE
Call-ID: Y2M1ODQxNTZmMjdkZWZjN2U5MmMyYjBmN2Y2OGY1ODQ._leg2
Route: 
Max-Forwards: 69
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
User-Agent: Z 3.3.25608 r25552
Content-Type: application/sdp
Contact: 
Content-Length: 246
.
.
.
For this reason, Is there a way to avoid this issue?
Also, Why in Kamailio, there is not b2bua module so that perform all b2bua
functionality? i know it is possible to do this by using other modules like
textops and retransformation ?
Thanks with Best Regards.
Mojtaba

On Thu, Jul 8, 2021 at 5:49 PM Dmitry Sinina 
wrote:

> I mean there are no commits to this modules for years(excluding may be
> some cosmetic fixes) when yeti-switch sems core was changed a lot.
>
> On 08.07.21 15:59, Juha Heinanen wrote:
> > Dmitry Sinina writes:
> >
> >> Currently https://github.com/yeti-switch/sems supports zrtp as well as
> >> srtp-dtls and srtp-sdes. But you are right, old applications like sbc
> >> and dsm requires modifications to run.
> > What do you mean by "old"?  sbc and dsm apps are much younger that many
> > others.
> >
> > -- Juha
> >
> > __
> > Kamailio - Users Mailing List - Non Commercial Discussions
> >* sr-users@lists.kamailio.org
> > Important: keep the mailing list in the recipients, do not reply only to
> the sender!
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Re: [SR-Users] Question about Kamailio IMS in box

2021-05-17 Thread Mojtaba
absolutely yes.
Just set local DNS ip server in nslookup file in Ubuntu.
With regards.

On Wed, 12 May 2021, 20:48 Ibrahim soumana djibo, 
wrote:

> Hello,
>
> I am new to kamailio. And I Am following the work of Franz Edler.
>
> I would like to test it in a local environment without internet connection
> on ubuntu. What I don't understand is, to download the packages I
> necessarily need the internet. Do I have to use two network interfaces for
> the same virtual machine in order to access the internet and make the DNS
> configuration locally with the second interface?
>
> Cordially,
>
>
>
> *SOUMANA** DJIBO **Ibrahim*
> *Ingénieur Réseaux & Télécoms*
> *5 Avenue Emile Baudot, 91300 Massy*
> +33 07 61 80 20 35
> [image: Résultat de recherche d'images pour "logo paris saclay"]
> https://www.universite-paris-saclay.fr/
>
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Re: [SR-Users] Can Kamailio help with this?

2021-05-15 Thread Mojtaba
Absolutely yes, In Kamailio with specific module you could do it. Some
related module for doing this scenario are:
presence, presence_xml, presence_dialoginfo, pua, pua_dialoginfo
You could know more about them by referring to its documentation.
With Best Regards

Mojtaba Esfandiari.S

On Sat, May 15, 2021 at 5:52 PM Antony Stone <
antony.st...@kamailio.open.source.it> wrote:

> Hi.
>
> I've been aware of Kamailio, and on this list, for several years, but so
> far I
> have not implemented Kamailio for any purpose.
>
> I wonder if it could be a suitable tool for the following scenario.
>
> I currently use Asterisk to register as a client, with username and
> password,
> to other SIP PBXs in order to receive calls.  As far as the other PBX is
> concerned, Asterisk looks like a SIP telephone on a particular extension.
>
> However, Asterisk's SIP client capabilities are limited to handling phone
> calls only, and it specifically cannot receive presence information about
> the
> state of other extensions from the remote PBX.
>
> So, could Kamailio be used in the path between Asterisk and the remote PBX
> so
> that the registration on the PBX comes from Kamailio (with any incoming
> calls
> being passed to Asterisk), but with Kamailio also receiving presence
> information from the remote PBX and making this available to some script
> or
> application (which I would expect to have to write)?
>
> The main point is that I don't want to have two things registering to the
> remote PBX, one for calls and one for presence, so I'm looking for a way
> to
> register with something which understands presence, and can also pass
> calls on
> to Asterisk.
>
> Can anyone suggest whether Kamailio could do this, and if so, point me at
> some
> resources to help me get started; or alternatively suggest ideas on some
> other
> tool which might be more appropriate than Kamailio?
>
>
> Thanks for any ideas,
>
>
> Antony.
>
> --
> Schrödinger's rule of data integrity: the condition of any backup is
> unknown
> until a restore is attempted.
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
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Re: [SR-Users] connect multiple asterisk to kamailio

2021-05-03 Thread Mojtaba
Hello Vahid,
Ofcourse, you could use multi-domain feature in Kamailio. Your customers
could put in specific domain like as sip:custom...@domain1.com or
sip:customer2
@domain2.com.
All customers have to send their requests to kamailio which is defined in
outbound-proxy field. In kamailio, based on their domains, the requests
will relay to their asterisk or upstream servers.
Thanks.
Mojtaba Esfandiari.S

On Fri, Apr 30, 2021 at 11:31 PM Vahid sadrnia 
wrote:

> hi everybody
> how can i connect multiple asterisk to kamailio and user just connect to
> specific asterisk(not load balancing),in fact each customer have their own
> media gateway(asterisk) and use kamailio for topolgy hiding.
> by regards and thanks.
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Re: [SR-Users] No private identity specified

2021-04-08 Thread Mojtaba
Please attach pcap file of signaling flow in each nodes.

On Thu, Apr 8, 2021 at 2:07 PM Dong Ming (董明)  wrote:

> Thank you for your reply!
>
>
>
> It's the first time to learn ims, I am very sorry for lack of knowledge
> and inaccurate description of this problem.
>
> I sincerely hope that you can give me some advice how to deal with this
> problem.
>
> thank you!
>
>
>
> *发件人:* Mojtaba 
> *发送时间:* 2021年4月8日 17:02
> *收件人:* Dong Ming (董明) 
> *抄送:* Daniel-Constantin Mierla ; Kamailio (SER) -
> Users Mailing List 
> *主题:* Re: [SR-Users] sip registration fail
>
>
>
> Just be notic, being behind nat, cause your request to pcscf is rejected
> and the same issue (No private identity specified) is appeared in lon
> info.
>
> The public identity (IMPU) and private identity (IMPI) are created with
> the same pattern by default as shown in your attachment, but the title of
> the issue is confusing.
>
>
>
>
>
> On Thu, Apr 8, 2021 at 1:08 PM Dong Ming (董明)  wrote:
>
> Hello
>
>
>
> According to the debug information,
>
> 
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:610]: parse_msg(): SIP
> Request:
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:612]: parse_msg():
> method:  
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:614]: parse_msg():
> uri: 
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:616]: parse_msg():
> version: 
>
> 12(4516) DEBUG:  [core/parser/parse_addr_spec.c:864]:
> parse_addr_spec(): end of header reached, state=10
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:171]: get_hdr_field():
>  [57]; uri=[sip:46001000...@ims.mnc001.mcc460.3gppnetwork.org]
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:174]: get_hdr_field(): to
> body [
>
> ], to tag []
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:152]: get_hdr_field():
> cseq : <2> 
>
> 12(4516) DEBUG:  [core/parser/parse_via.c:1303]: parse_via_param():
> Found param type 232,  =
> ; state=6
>
> 12(4516) DEBUG:  [core/parser/parse_via.c:1303]: parse_via_param():
> Found param type 236,  = <1>; state=16
>
> 12(4516) DEBUG:  [core/parser/parse_via.c:2639]: parse_via(): end of
> header reached, state=5
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:498]: parse_headers():
> Via found, flags=2
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:500]: parse_headers():
> this is the first via
>
> 12(4516) INFO: ims_auth [cxdx_avp.c:137]: cxdx_get_avp():
> cxdx_get_user_name: Failed finding avp (avp_code = 1, vendor_id = 0)
>
> 12(4516) ERROR: ims_auth [cxdx_mar.c:151]: async_cdp_callback(): No
> private identity specified (Authorization: username)
>
> ====
>
> The above content is found in the log of s-cscf,It seems that the
> authentication failed.
>
> May I ask how to configure this part, or if this part of the code is
> deleted, will it affect the subsequent process?
>
>
>
> Thank you very much!
>
>
>
> *发件人:* Mojtaba 
> *发送时间:* 2021年4月8日 15:48
> *收件人:* Daniel-Constantin Mierla ; Kamailio (SER) -
> Users Mailing List 
> *抄送:* Dong Ming (董明) 
> *主题:* Re: [SR-Users] sip registration fail
>
>
>
> It seems the issue is related to some default configuration in IMS nodes
> in(pcscf, icscf and scscf). In pcap file, just pcscf node is appeared.
>
> please send full signaling flow in whole nodes.
>
> Thanks
>
>
>
> On Thu, Apr 8, 2021 at 11:06 AM Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
> Hello,
>
> On 07.04.21 10:40, Dong Ming (董明) wrote:
>
> Hello
>
>
>
> I have a problem about volte(401 Unauthorized).
>
>
>
> *P-CSCF:*
>
> 4(4467) INFO: rr [rr_mod.c:515]: pv_get_route_uri_f(): No route header
> present.
> 4(4467) INFO: ims_registrar_pcscf [sec_agree.c:296]:
> cscf_get_security_verify(): No sec urity-verify parameters found
>
>
>
> Can you help me see where the problem is?
>
> Tips:About SQN, the SQN is the same at the beginning of the test
> (Open5GS, FOHSS), but when it is found that the sip registration fails, the
> SQN is inconsistent. If this situation is normal? If it is not normal, how
> to solve it?
>
> I am not familiar with the IMS extensions, if nobody else comes with other
> suggestions, I recommend to set debug=3 in your kamailio configuration file
> and then look at the syslog, there should be a lot of DEBUG messages that
> may give more hints about what happens.
>
> Also, when sending Kamailio logs, do not make screen shots, just copy and
> paste them, it is easier to read as well as search in the source code for
> them by

Re: [SR-Users] sip registration fail

2021-04-08 Thread Mojtaba
Just be notic, being behind nat, cause your request to pcscf is rejected
and the same issue (No private identity specified) is appeared in lon info.
The public identity (IMPU) and private identity (IMPI) are created with the
same pattern by default as shown in your attachment, but the title of the
issue is confusing.


On Thu, Apr 8, 2021 at 1:08 PM Dong Ming (董明)  wrote:

> Hello
>
>
>
> According to the debug information,
>
> 
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:610]: parse_msg(): SIP
> Request:
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:612]: parse_msg():
> method:  
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:614]: parse_msg():
> uri: 
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:616]: parse_msg():
> version: 
>
> 12(4516) DEBUG:  [core/parser/parse_addr_spec.c:864]:
> parse_addr_spec(): end of header reached, state=10
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:171]: get_hdr_field():
>  [57]; uri=[sip:46001000...@ims.mnc001.mcc460.3gppnetwork.org]
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:174]: get_hdr_field(): to
> body [
>
> ], to tag []
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:152]: get_hdr_field():
> cseq : <2> 
>
> 12(4516) DEBUG:  [core/parser/parse_via.c:1303]: parse_via_param():
> Found param type 232,  =
> ; state=6
>
> 12(4516) DEBUG:  [core/parser/parse_via.c:1303]: parse_via_param():
> Found param type 236,  = <1>; state=16
>
> 12(4516) DEBUG:  [core/parser/parse_via.c:2639]: parse_via(): end of
> header reached, state=5
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:498]: parse_headers():
> Via found, flags=2
>
> 12(4516) DEBUG:  [core/parser/msg_parser.c:500]: parse_headers():
> this is the first via
>
> 12(4516) INFO: ims_auth [cxdx_avp.c:137]: cxdx_get_avp():
> cxdx_get_user_name: Failed finding avp (avp_code = 1, vendor_id = 0)
>
> 12(4516) ERROR: ims_auth [cxdx_mar.c:151]: async_cdp_callback(): No
> private identity specified (Authorization: username)
>
> 
>
> The above content is found in the log of s-cscf,It seems that the
> authentication failed.
>
> May I ask how to configure this part, or if this part of the code is
> deleted, will it affect the subsequent process?
>
>
>
> Thank you very much!
>
>
>
> *发件人:* Mojtaba 
> *发送时间:* 2021年4月8日 15:48
> *收件人:* Daniel-Constantin Mierla ; Kamailio (SER) -
> Users Mailing List 
> *抄送:* Dong Ming (董明) 
> *主题:* Re: [SR-Users] sip registration fail
>
>
>
> It seems the issue is related to some default configuration in IMS nodes
> in(pcscf, icscf and scscf). In pcap file, just pcscf node is appeared.
>
> please send full signaling flow in whole nodes.
>
> Thanks
>
>
>
> On Thu, Apr 8, 2021 at 11:06 AM Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
> Hello,
>
> On 07.04.21 10:40, Dong Ming (董明) wrote:
>
> Hello
>
>
>
> I have a problem about volte(401 Unauthorized).
>
>
>
> *P-CSCF:*
>
> 4(4467) INFO: rr [rr_mod.c:515]: pv_get_route_uri_f(): No route header
> present.
> 4(4467) INFO: ims_registrar_pcscf [sec_agree.c:296]:
> cscf_get_security_verify(): No sec urity-verify parameters found
>
>
>
> Can you help me see where the problem is?
>
> Tips:About SQN, the SQN is the same at the beginning of the test
> (Open5GS, FOHSS), but when it is found that the sip registration fails, the
> SQN is inconsistent. If this situation is normal? If it is not normal, how
> to solve it?
>
> I am not familiar with the IMS extensions, if nobody else comes with other
> suggestions, I recommend to set debug=3 in your kamailio configuration file
> and then look at the syslog, there should be a lot of DEBUG messages that
> may give more hints about what happens.
>
> Also, when sending Kamailio logs, do not make screen shots, just copy and
> paste them, it is easier to read as well as search in the source code for
> them by copy
>
> Cheers,
> Daniel
>
>
>
> --
>
> Daniel-Constantin Mierla -- www.asipto.com
>
> www.twitter.com/miconda -- www.linkedin.com/in/miconda
>
> Kamailio Advanced Training - Online
>
> May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone)
>
>   * https://www.asipto.com/sw/kamailio-advanced-training-online/
>
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>
>
>
> --
>
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> 本电子邮件(包括任何的附件)为本公司保密文件。本文件仅仅可为

Re: [SR-Users] sip registration fail

2021-04-08 Thread Mojtaba
It seems the issue is related to some default configuration in IMS nodes
in(pcscf, icscf and scscf). In pcap file, just pcscf node is appeared.
please send full signaling flow in whole nodes.
Thanks

On Thu, Apr 8, 2021 at 11:06 AM Daniel-Constantin Mierla 
wrote:

> Hello,
> On 07.04.21 10:40, Dong Ming (董明) wrote:
>
> Hello
>
>
>
> I have a problem about volte(401 Unauthorized).
>
>
>
> *P-CSCF:*
>
> 4(4467) INFO: rr [rr_mod.c:515]: pv_get_route_uri_f(): No route header
> present.
> 4(4467) INFO: ims_registrar_pcscf [sec_agree.c:296]:
> cscf_get_security_verify(): No sec urity-verify parameters found
>
>
>
> Can you help me see where the problem is?
>
> Tips:About SQN, the SQN is the same at the beginning of the test
> (Open5GS, FOHSS), but when it is found that the sip registration fails, the
> SQN is inconsistent. If this situation is normal? If it is not normal, how
> to solve it?
>
> I am not familiar with the IMS extensions, if nobody else comes with other
> suggestions, I recommend to set debug=3 in your kamailio configuration file
> and then look at the syslog, there should be a lot of DEBUG messages that
> may give more hints about what happens.
>
> Also, when sending Kamailio logs, do not make screen shots, just copy and
> paste them, it is easier to read as well as search in the source code for
> them by copy
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 
> www.linkedin.com/in/miconda
> Kamailio Advanced Training - Online
> May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone)
>   * https://www.asipto.com/sw/kamailio-advanced-training-online/
>
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Re: [SR-Users] Next online devel meeting - Wed, Nov 25, 2020, 15:00UTC

2020-11-17 Thread Mojtaba
Hello Daniel, How are you?
I want to join in online devel meeting for talking a littte bit about
LREProxy module. I hope it could be release for Kamailio V5.5.
Is it possible for doing this?
With Best Regards.
Mojtaba

On Mon, Nov 16, 2020 at 4:01 PM Daniel-Constantin Mierla 
wrote:

> Hello,
>
> in order to plan the roadmap to next major release, it is time to
> organize another online devel meeting, so the developers and community
> members can syncronize and plan what should be done till Kamailio v5.5
> will be out.
>
> Because freenode.net service is requiring user registration to join
> IRC channels on their servers, the meeting will be done on a matrix chat
> room. More details are in the wiki page of the event, available at:
>
>   * https://www.kamailio.org/wiki/devel/irc-meetings/2020b
>
> Feel free to add there topics that you want to be discussed.
>
> The proposed date is next Wednesday, November 25, 2020, at 15:00UTC (15:00
> London/Dublin, 16:00 most of West Europe, 10:00 New York), but other
> dates can be proposed if they suit for more participants.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla -- www.asipto.com
> www.twitter.com/miconda -- www.linkedin.com/in/miconda
>
>
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Re: [SR-Users] Question about textops

2020-06-26 Thread Mojtaba
Any idea?

On Wed, 24 Jun 2020, 14:26 Mojtaba,  wrote:

> Hello,
> Some confusing things while developing in Kamailio:
> In is_method_f function in textops module, the code checks the HDR_CSEQ_F
> in msg also, and if the method name appears in CSEQ_F, The result is
> returned TRUE.
> These question will  arise why in this function, the CSEQ field in message
> will check?
> if(parse_headers(msg, HDR_CSEQ_F, 0)!=0 || msg->cseq==NULL)
> {
> LM_ERR("cannot parse cseq header\n");
> return -1; /* should it be 0 ?!?! */
> }
> if(m->s==0)
> return (get_cseq(msg)->method_id>len)?1:-1;
> else
> return (get_cseq(msg)->method_id==METHOD_OTHER
> && get_cseq(msg)->method.len==m->len
> && (strncasecmp(get_cseq(msg)->method.s, m->s,
> m->len)==0))?1:-1;
>
> I think it would be nice to add new function like: is_cseq_method_f in
> module to avoid some problem understanding.
>
> --
> --Mojtaba Esfandiari.S
>
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[SR-Users] Question about textops

2020-06-25 Thread Mojtaba
Hello,
Some confusing things while developing in Kamailio:
In is_method_f function in textops module, the code checks the HDR_CSEQ_F
in msg also, and if the method name appears in CSEQ_F, The result is
returned TRUE.
These question will  arise why in this function, the CSEQ field in message
will check?
if(parse_headers(msg, HDR_CSEQ_F, 0)!=0 || msg->cseq==NULL)
{
LM_ERR("cannot parse cseq header\n");
return -1; /* should it be 0 ?!?! */
}
if(m->s==0)
return (get_cseq(msg)->method_id>len)?1:-1;
else
return (get_cseq(msg)->method_id==METHOD_OTHER
&& get_cseq(msg)->method.len==m->len
&& (strncasecmp(get_cseq(msg)->method.s, m->s,
m->len)==0))?1:-1;

I think it would be nice to add new function like: is_cseq_method_f in
module to avoid some problem understanding.

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[SR-Users] Free Persian Asterisk Book and COVID-19 Pandemic

2020-04-19 Thread Mojtaba
In the context of the COVID-19 pandemic, Nasim Telecom decided to publish
the first six-captures of Persian Asterisk book that was written by me and
Mr Najafi, published in 2017.
Also it was introduced by Mr. David Duffett in AstriCon 2017 on Florida.
It would be useful for all people which understand Persian language, Also
all other people could use it by going code commands on it and examples.

For downloading it go the following link:

https://www.nasimtelecom.com/en/persian-asterisk-book/

So, be careful and safety at first, Stay at home and read your favourite
books.
With Best Regards.
  --Mojtaba Esfandiari.S
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Re: [SR-Users] RTPEngine performance - How far are you pushing your systems

2020-01-07 Thread Mojtaba
Absolutely using wireshark to capture all RTP traffic for analysing is very
challengeable because the size of capture file is very huge. So i have used
-C option to spilite to multiple pcap file. I shared a picture of it:
https://www.linkedin.com/posts/mojtaba-esfandiari_kamailio-kamailioworld-activity-6605747231586947072-M_x9

The result of you get is strange!

On Tue, Jan 7, 2020 at 1:24 PM davy van de moere <
davy.van.de.mo...@gmail.com> wrote:

> Using wireshark, I get a jitter still smaller than 1ms. You have efficient
> ways to measure that?
>
> Op di 7 jan. 2020 om 10:43 schreef Mojtaba :
>
>> Yes, I reached 3000 channels in kernel mode with about 25% usage of one
>> core. Its incredible. But be notic the number of channels is not matter,
>> What about its QoS parameter such as max-delta, jitter, lost?
>> The max delay i have with the same result is 60.
>>
>> On Tue, Jan 7, 2020 at 12:56 PM davy van de moere <
>> davy.van.de.mo...@gmail.com> wrote:
>>
>>> As an update, I'm getting 2000 channels in kernel mode, with about 50%
>>> usage of 1 core. (load of the server is at 0.5), which is a 10 fold
>>> difference.
>>>
>>> The problem I was hitting, was the (probable) inability to work in
>>> kernelmode with the rtpproxy.so module, rtpengine.so works nicely.
>>>
>>> Op di 7 jan. 2020 om 10:19 schreef Mojtaba :
>>>
>>>> As a person who working much on developing new RTP engine for Kamailio,
>>>> I should express achieving to 12k concurrent call is very strange with
>>>> Kernel part of RTPEngine.
>>>> Regardless of NIC's parameter, multi core and some other parameters, it
>>>> is important where the hooking linux kernel function is called. The Kernel
>>>> part of RTPEngine is called in NF_INET_LOCAL_IN. It means the rtp packet
>>>> goes up local routing in network stack.
>>>> I could not reach to 12k yet, although i call Linux kernel function in
>>>> NF-INET-PRE-ROUTING
>>>>
>>>> On Mon, Jan 6, 2020 at 8:26 PM davy van de moere <
>>>> davy.van.de.mo...@gmail.com> wrote:
>>>>
>>>>> It will probably be a case of RTFM...
>>>>>
>>>>> quoting from: https://github.com/sipwise/rtpengine/blob/master/README.md
>>>>>
>>>>> # ensure that the table we want to use doesn't exist - usually needed 
>>>>> after a daemon
>>>>> # restart, otherwise will error
>>>>> echo 'del 0' > /proc/rtpengine/control
>>>>>
>>>>>
>>>>> this part seems to be very relevant indeed!
>>>>>
>>>>> Op ma 6 jan. 2020 om 17:41 schreef Nicolas Breuer <
>>>>> nicolas.bre...@belcenter.biz>:
>>>>>
>>>>>> Hey Davy,
>>>>>>
>>>>>>
>>>>>>
>>>>>> Check  cat /proc/rtpengine/0/list
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> *De :* sr-users  *De la part de*
>>>>>> davy van de moere
>>>>>> *Envoyé :* lundi 6 janvier 2020 14:47
>>>>>> *À :* Henning Westerholt 
>>>>>> *Cc :* Kamailio (SER) - Users Mailing List <
>>>>>> sr-users@lists.kamailio.org>
>>>>>> *Objet :* Re: [SR-Users] RTPEngine performance - How far are you
>>>>>> pushing your systems
>>>>>>
>>>>>>
>>>>>>
>>>>>> Ok, I'll get my SIPP warmed up then ... ;) thx for the feedback!
>>>>>>
>>>>>>
>>>>>>
>>>>>> Op ma 6 jan. 2020 om 14:45 schreef Henning Westerholt >>>>> >:
>>>>>>
>>>>>> Hello Davy,
>>>>>>
>>>>>>
>>>>>>
>>>>>> You should look into it, sounds strange. Compare e.g. to this e-mail
>>>>>> from 2015 – over 16k simultaneous calls on a 12 core machine:
>>>>>>
>>>>>> https://lists.kamailio.org/pipermail/sr-dev/2015-October/031432.html
>>>>>>
>>>>>>
>>>>>>
>>>>>> Cheers,
>>>>>>
>>>>>>
>>>>>>
>>>>>> Henning
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>

Re: [SR-Users] RTPEngine performance - How far are you pushing your systems

2020-01-07 Thread Mojtaba
Yes, I reached 3000 channels in kernel mode with about 25% usage of one
core. Its incredible. But be notic the number of channels is not matter,
What about its QoS parameter such as max-delta, jitter, lost?
The max delay i have with the same result is 60.

On Tue, Jan 7, 2020 at 12:56 PM davy van de moere <
davy.van.de.mo...@gmail.com> wrote:

> As an update, I'm getting 2000 channels in kernel mode, with about 50%
> usage of 1 core. (load of the server is at 0.5), which is a 10 fold
> difference.
>
> The problem I was hitting, was the (probable) inability to work in
> kernelmode with the rtpproxy.so module, rtpengine.so works nicely.
>
> Op di 7 jan. 2020 om 10:19 schreef Mojtaba :
>
>> As a person who working much on developing new RTP engine for Kamailio, I
>> should express achieving to 12k concurrent call is very strange with Kernel
>> part of RTPEngine.
>> Regardless of NIC's parameter, multi core and some other parameters, it
>> is important where the hooking linux kernel function is called. The Kernel
>> part of RTPEngine is called in NF_INET_LOCAL_IN. It means the rtp packet
>> goes up local routing in network stack.
>> I could not reach to 12k yet, although i call Linux kernel function in
>> NF-INET-PRE-ROUTING
>>
>> On Mon, Jan 6, 2020 at 8:26 PM davy van de moere <
>> davy.van.de.mo...@gmail.com> wrote:
>>
>>> It will probably be a case of RTFM...
>>>
>>> quoting from: https://github.com/sipwise/rtpengine/blob/master/README.md
>>>
>>> # ensure that the table we want to use doesn't exist - usually needed after 
>>> a daemon
>>> # restart, otherwise will error
>>> echo 'del 0' > /proc/rtpengine/control
>>>
>>>
>>> this part seems to be very relevant indeed!
>>>
>>> Op ma 6 jan. 2020 om 17:41 schreef Nicolas Breuer <
>>> nicolas.bre...@belcenter.biz>:
>>>
>>>> Hey Davy,
>>>>
>>>>
>>>>
>>>> Check  cat /proc/rtpengine/0/list
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> *De :* sr-users  *De la part de*
>>>> davy van de moere
>>>> *Envoyé :* lundi 6 janvier 2020 14:47
>>>> *À :* Henning Westerholt 
>>>> *Cc :* Kamailio (SER) - Users Mailing List >>> >
>>>> *Objet :* Re: [SR-Users] RTPEngine performance - How far are you
>>>> pushing your systems
>>>>
>>>>
>>>>
>>>> Ok, I'll get my SIPP warmed up then ... ;) thx for the feedback!
>>>>
>>>>
>>>>
>>>> Op ma 6 jan. 2020 om 14:45 schreef Henning Westerholt :
>>>>
>>>> Hello Davy,
>>>>
>>>>
>>>>
>>>> You should look into it, sounds strange. Compare e.g. to this e-mail
>>>> from 2015 – over 16k simultaneous calls on a 12 core machine:
>>>>
>>>> https://lists.kamailio.org/pipermail/sr-dev/2015-October/031432.html
>>>>
>>>>
>>>>
>>>> Cheers,
>>>>
>>>>
>>>>
>>>> Henning
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> Henning Westerholt – https://skalatan.de/blog/
>>>>
>>>> Kamailio services – https://gilawa.com
>>>>
>>>>
>>>>
>>>> *From:* sr-users  *On Behalf Of *davy
>>>> van de moere
>>>> *Sent:* Monday, January 6, 2020 12:52 PM
>>>> *To:* Kamailio (SER) - Users Mailing List 
>>>> *Subject:* [SR-Users] RTPEngine performance - How far are you pushing
>>>> your systems
>>>>
>>>>
>>>>
>>>> Dear all,
>>>>
>>>>
>>>>
>>>> Not a direct question on kamailio, but on rtpengine, so feel free to
>>>> ignore :)
>>>>
>>>>
>>>>
>>>> I have a farm of rtpengine servers , supporting our set of kamailio's.
>>>>
>>>>
>>>>
>>>> On recent decent hardware (hp dl360 gen9/10 servers), I typically push
>>>> my systems upto 2500 channels per machine. Then I tend to see the load of
>>>> the machine go up in a logarithmic manner. E.g. a load of 3 supports 2500
>>>> channels, but when its at 3000 channels, load can be at 8. etc.
>>>>
>>>>
>>>>
>>>> I have the kernel module running, and I have a good impression it's
>>>

Re: [SR-Users] Need support

2019-12-20 Thread Mojtaba
Hello,
In regard of performance testing of your product, both performance &
conformance i have shared my experience in twitter in the following:
https://twitter.com/MespioS/status/1183303415801221120
https://twitter.com/MespioS/status/1199973639120195584
What type of performance test (stress test) you are looking for?

On Wed, Dec 11, 2019 at 10:48 AM Srikanth Magloori <
srikanth.maglo...@valuelabs.com> wrote:

> Hi Team,
>
> We are using your product Kamalio in Call Recording services.
>
> We are planning for Performance testing of our recording system.
>
> Currently we have bypassed Twilio and Kamailio.
>
> Generating Calls using SIPP directly to VOIP.
>
> So, we want calls to route through KAMAILIO -> VOIP. So performance can be
> bit realistic.
>
> Any help is greatly appreciated!
>
> Thanks
> Srikanth
>
>
>
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Re: [SR-Users] kamailio IMS supporting VOLTE

2019-12-04 Thread Mojtaba
Absolutely yes,
Let me clear the answer. The Kamailio has some module to support some
Diameter interface such as Cx, Ro and ext, but it has not built-in HSS. You
have to use FHOSS or develop your own HSS. Fortunately because of existing
IMS-Diameter module in Kamailio, it would be easily to develop your own
HSS without no more knowledge about Diameter protocol.
With best regards.
Mojtaba

On Wed, Dec 4, 2019 at 10:39 AM Shubhendu Das 
wrote:

>
>
> Hi ,
>
>
>
> Does Kamailio has  its own HSS supporting CX Interface ?
>
>
>
> BR,
>
> Shubhendu
>
>
>
> *From:* sr-users [mailto:sr-users-boun...@lists.kamailio.org] *On Behalf
> Of *Mojtaba
> *Sent:* Saturday, November 23, 2019 4:37 PM
> *To:* Kamailio (SER) - Users Mailing List
> *Subject:* Re: [SR-Users] kamailio IMS supporting VOLTE
>
>
>
> Hello,
>
> Running the IMS network for supporting VoLTE with Kamailio module and
> FHOSS is challenging. THe Kamailio works great in IMS network but You have
> to optimise your own database. Using FHOSS without any support or
> development  is very risky.
>
> In general speaking, running VoLTE have three main section and the IMS
> network is one of them. You might have some challenges in EPC and RAN
> access.
>
> WIth Best Regards.
>
> Mojtaba
>
>
>
>
>
> On Thu, Nov 21, 2019 at 3:23 PM Shubhendu Das 
> wrote:
>
> Hi ,
>
>
>
> I have a query regarding Kamailio open source SIP server.
>
> Does it have any release, where its supports the feature of IMS supporting
> VOLTE with real commercial mobile phones(IMSI based) ?
>
>
>
> If so, what is the release number ? Please also let me know about the
> supported mobile phones list.
>
>
>
> BR,
>
> Shubhendu
>
>
>
>
> *Disclaimer: This email and any files transmitted along with it may
> contain Azcom confidential and proprietary information.  If you are not the
> intended recipient, you are notified that disclosing, copying, distributing
> or taking any action based on the contents of the information contained
> herein is strictly prohibited. If you are not an intended recipient of this
> transmission and you received it in error, please inform the sender by
> reply e-mail and destroy this and all other copies of this transmission to
> which you have access. The sender of this email or Azcom does not accept
> liability for any errors or omissions in the contents of this message that
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>
>
>
> --
>
> --Mojtaba Esfandiari.S
>
> *Disclaimer: This email and any files transmitted along with it may
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Re: [SR-Users] tcp limit, too many open files

2019-11-28 Thread Mojtaba
Hello,
You can use ulimit command with -n cimmand to set max open file before
start Kamailio.

On Thu, 28 Nov 2019, 18:52 David Villasmil, 
wrote:

> Thanks Alex,
>
> Do you know how to set it? First time doing thousands of tcp
> registrations..
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Thu, Nov 28, 2019 at 3:16 PM Alex Balashov 
> wrote:
>
>> There is a userspace open file descriptor limit you need to increase for
>> the user under which Kamailio is running.
>>
>> —
>> Sent from mobile, with due apologies for brevity and errors.
>>
>> On Nov 28, 2019, at 10:08 AM, David Villasmil <
>> david.villasmil.w...@gmail.com> wrote:
>>
>> 
>> Hello guys,
>>
>> I'm facing an issue where, if i start kamailio as root, all works
>> properly, but if i start it as kamailio, i get hundreds of:
>>
>> handle_new_connect(): error while accepting connection(24): Too many open
>> files
>>
>> My limits is like:
>>
>> kamailio   softcoreunlimited
>> kamailio   softdataunlimited
>> kamailio   softfsize   unlimited
>> kamailio   softmemlock unlimited
>> kamailio   softnofile  unlimited
>> kamailio   softrss unlimited
>> kamailio   hardstack   240
>> kamailio   softcpu unlimited
>> kamailio   softnproc   unlimited
>> kamailio   softas  unlimited
>> kamailio   softpriority-11
>> kamailio   softlocks   unlimited
>> kamailio   softsigpending  unlimited
>> kamailio   softmsgqueueunlimited
>> kamailio   softnice-11
>> kamailio   softnofile  65536
>> kamailio   hardnofile  65536
>>
>> root   softcoreunlimited
>> root   softdataunlimited
>> root   softfsize   unlimited
>> root   softmemlock unlimited
>> root   softnofile  unlimited
>> root   softrss unlimited
>> root   hardstack   240
>> root   softcpu unlimited
>> root   softnproc   unlimited
>> root   softas  unlimited
>> root   softpriority-11
>> root   softlocks   unlimited
>> root   softsigpending  unlimited
>> root   softmsgqueueunlimited
>> root   softnice-11
>> root   softnofile  65536
>> root   hardnofile  65536
>>
>> I can't figure out why this is happening...
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.w...@gmail.com
>> phone: +34669448337
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Re: [SR-Users] kamailio IMS supporting VOLTE

2019-11-23 Thread Mojtaba
Hello,
Running the IMS network for supporting VoLTE with Kamailio module and FHOSS
is challenging. THe Kamailio works great in IMS network but You have to
optimise your own database. Using FHOSS without any support or development
is very risky.
In general speaking, running VoLTE have three main section and the IMS
network is one of them. You might have some challenges in EPC and RAN
access.
WIth Best Regards.
Mojtaba


On Thu, Nov 21, 2019 at 3:23 PM Shubhendu Das 
wrote:

> Hi ,
>
>
>
> I have a query regarding Kamailio open source SIP server.
>
> Does it have any release, where its supports the feature of IMS supporting
> VOLTE with real commercial mobile phones(IMSI based) ?
>
>
>
> If so, what is the release number ? Please also let me know about the
> supported mobile phones list.
>
>
>
> BR,
>
> Shubhendu
>
>
>
> *Disclaimer: This email and any files transmitted along with it may
> contain Azcom confidential and proprietary information.  If you are not the
> intended recipient, you are notified that disclosing, copying, distributing
> or taking any action based on the contents of the information contained
> herein is strictly prohibited. If you are not an intended recipient of this
> transmission and you received it in error, please inform the sender by
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Re: [SR-Users] IMS Kamailio Diameter Charging Server

2019-10-12 Thread Mojtaba
Hello,
You have to enable ims_charging control function in your network element
(SCSCF), then connect  it to Online Charging Server Module with ims_ocs
module.
Be notice about it's FQDN of each module.
With Best Regards.

On Fri, Oct 11, 2019 at 5:40 PM Jack R  wrote:

> Hi All,
>
> We have installed Kamailio as IMS setup pcscf icscf scscf - 3 instances
> running on Ubuntu..
>
> Referring to the link below, it mentioned about Diameter based Kamailio
> Charging Server.
>
> https://www.kamailio.org/events/2016-KamailioWorld/Day2/15-Carsten.Bock-Diameter-Charging-With-Kamailio.pdf
>
>
> Where to find it? How to install it? to make it work in Kamailio IMS set
> up as per below articles:
> https://www.kamailio.org/docs/modules/stable/modules/ims_charging.html
> https://kamailio.org/docs/modules/devel/modules/ims_ocs.html
>
> Regards,
> Jack.
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>


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Re: [SR-Users] Relay Request

2019-09-29 Thread Mojtaba
Would you please keep "Users Mailing List"  in
tracking. :)
That is straight froward. There is no change in sending the request from
Kamailio to Asterisk. In Asterisk you could use Progress command to support
Media-Relay. In additional use Dial command (with r or m option for
supporting CRBT ).
With Best Regards

On Sun, Sep 29, 2019 at 8:52 AM Maryam Baghdadi 
wrote:

> Hello Mojtaba,
> I chose asterisk as media server and some services like voice mail
> provided ... about CRBT I know asterisk have this service but I don't know
> how to send this request from kamailio to asterisk..
> do you know kamailio can provide this service itself or not???
> could you pls explain more
>
> thanks
>
>
> On Sat, Sep 28, 2019, 1:48 PM Mojtaba 
>> Hello, Some consideration would be needed:
>> 1) SS7 for PSTN subscriber to support media-relay
>> 2)Progress capability  for SIP subscriber.
>> 3) Media server to support ring-back VAS service
>> You should check enabling this service or not during MO procedure.
>> With Best Regards.
>>
>> On Sat, Sep 28, 2019 at 1:19 PM Maryam Baghdadi <
>> maryam.baghd...@gmail.com> wrote:
>>
>>> Hello everyone,
>>>
>>> Is it possible to relay INVITE request to another server after receiving
>>> "180 Ringing" response ???
>>> I would like to activate CRBT(ring back tone) service and according to
>>> my research after receiving 180 ringing media playing request should be
>>> sent to a media server .
>>> _______
>>> 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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Re: [SR-Users] [sr-dev] IMS_ISC technical specification

2019-07-24 Thread Mojtaba
According to TS 24.229, The AS could send request to SCSCF, too.
But some rules shall be obeied.
For example did you add P-Asserted-Identity header to your request?
With Best Regards.Mojtaba

On Wed, 24 Jul 2019, 17:19 Ali Shirvani  Hello Mojtaba,
>
> Thanks for your reply. You are completely correct, my current
> configuration work fine when request initiated from UEs, but I want to
> handle SIP requests that initiated by application server.
> When UE sends request to S-CSCF, kamailio fetch iFC from HSS and check if
> the request match with iFC or not, if it matched kamailio forward the
> request to the application server.
> But when application server send request to S-CSCF, isc_from_as('orig')
> return false and I couldn't forward request to the target UE.
>
> Regards,
> Ali
>
> On Wed, Jul 24, 2019 at 3:45 PM Mojtaba  wrote:
>
>> Hello Ali,
>> The SCSCF has IFC service for decision what it should do with incoming
>> SIP message. During check all rules in IFC, If it match with one or some
>> rules, actually it marked with that rules. Then the real meaning of marking
>> is the SIP message is matched with one or some rules in IFC.
>> With Best Regards.
>>
>> On Wed, Jul 24, 2019 at 2:48 PM Ali Shirvani 
>> wrote:
>>
>>> Hi all,
>>>
>>> I read some portion of imc_isc_mod.c, specially isc_from_as function. In
>>> that function there is some function call that searching for some marks in
>>> the SIP message. Also I read section 5.7.3 from TS 24.229 but I didn't find
>>> the marking mechanism that used in isc_from_as function.
>>>
>>> Would you please point me to the technical specification that
>>> implemented in IMS_ISC module?
>>>
>>> Regards,
>>> Ali
>>> ___
>>> Kamailio (SER) - Development Mailing List
>>> sr-...@lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev
>>>
>>
>>
>> --
>> --Mojtaba Esfandiari.S
>> ___
>> Kamailio (SER) - Development Mailing List
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>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev
>>
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Re: [SR-Users] [sr-dev] IMS_ISC technical specification

2019-07-24 Thread Mojtaba
Hello Ali,
The SCSCF has IFC service for decision what it should do with incoming SIP
message. During check all rules in IFC, If it match with one or some rules,
actually it marked with that rules. Then the real meaning of marking is the
SIP message is matched with one or some rules in IFC.
With Best Regards.

On Wed, Jul 24, 2019 at 2:48 PM Ali Shirvani  wrote:

> Hi all,
>
> I read some portion of imc_isc_mod.c, specially isc_from_as function. In
> that function there is some function call that searching for some marks in
> the SIP message. Also I read section 5.7.3 from TS 24.229 but I didn't find
> the marking mechanism that used in isc_from_as function.
>
> Would you please point me to the technical specification that implemented
> in IMS_ISC module?
>
> Regards,
> Ali
> ___
> Kamailio (SER) - Development Mailing List
> sr-...@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev
>


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Re: [SR-Users] kamailio as ICSCF connection with FHoss

2019-07-14 Thread Mojtaba
Sure, No Problem
Thank you


On Sun, Jul 14, 2019 at 1:22 PM Henning Westerholt  wrote:

> Hello Mojtaba,
>
> thank you. If there is something that should be added to the kamailio.cfg
> for this implementation (maybe some comments) just open a pull request.
>
> Cheers,
>
> Henning
> Am 14.07.19 um 10:28 schrieb Mojtaba:
>
> Hello Henning,
> According the 3GPP standards, the IMPU template would be started with '+',
> for this reason in icscf, we have to add '+' to $rU, But in FHOSS the is
> not any IMPU with '+'  prefix as default. There is two way to solve this
> issue:
> 1) Add some new IMPU/IMPI according the 3GPP standards. (Be notice that
> IMPU would be started with '+' or ' tel).
> 2) Have some changes in Kamailio routing block to escape some rules.
> The is no needed to have some changes in ims_icscf module.
>
> With Best Regards.Mojtaba
>
> On Sun, Jul 14, 2019 at 12:38 AM Henning Westerholt 
> wrote:
>
>> Hello Mojtaba,
>>
>> Let us wait for more details from the reporter before coming to a
>> conclusion. Without complete knowledge about the scenario there is always
>> some guessing involved.
>>
>> Cheers,
>>
>> Henning
>>
>> -Original Message-
>> From: sr-users  On Behalf Of Mojtaba
>> Sent: Saturday, July 13, 2019 8:36 PM
>> To: Kamailio (SER) - Users Mailing List ;
>> maryam.baghd...@gmail.com
>> Subject: Re: [SR-Users] kamailio as ICSCF connection with FHoss
>>
>> Hello Mariam,
>> It is worse way that you made choice. There is no needed to change any
>> things in source code.
>> Just dive to kamailio.cfg file and remove "+" prefix  from $rU (IMPU) .
>> It is very straightforward.
>> WIth Best Regards. Mojtaba
>>
>> On Sat, Jul 13, 2019 at 2:55 PM Maryam Baghdadi <
>> maryam.baghd...@gmail.com> wrote:
>> >
>> > Thank you for your help,
>> > I resolved my problem , there was a '+' in public_identity that isn't
>> shown in pcap file and just by printing value public_identity in
>> I_perform_location_information_request() recognized .
>> > I replaced  public_identity = cscf_get_public_identity_from_requri(msg)
>> with  cscf_get_to_uri(msg, _identity) then recompiled ims_icscf
>> module and my problem resolved .
>> >
>> >
>> >
>> > On Wed, Jul 10, 2019 at 3:07 PM Mojtaba  wrote:
>> >>
>> >> Hello,
>> >> of course not, Are you sure the IMPI/IMPU in LRI/LRA message there
>> >> are exist exactly in HSS?
>> >> Or may they are changed with some prefixes like "tel uri" or "+"
>> >> With Regards. Mojtaba
>> >>
>> >> On Wed, Jul 10, 2019 at 2:05 PM Henning Westerholt 
>> wrote:
>> >> >
>> >> > Hello Maryam,
>> >> >
>> >> >
>> >> >
>> >> > I am not an expert on the FHoss, but just some ideas how to
>> investigate it further:
>> >> >
>> >> >
>> >> >
>> >> > Do you get some error message in FHoss?
>> >> > Can you test with some other tool (maybe with a test script) that
>> the FHoss works ok generally?
>> >> > Any error message from ICSCF module in the Kamailio logging?
>> >> >
>> >> >
>> >> >
>> >> > Cheers,
>> >> >
>> >> >
>> >> >
>> >> > Henning
>> >> >
>> >> >
>> >> >
>> >> > --
>> >> >
>> >> > Henning Westerholt - https://skalatan.de/blog/
>> >> >
>> >> > Kamailio services - https://skalatan.de/services
>> >> >
>> >> >
>> >> >
>> >> > From: sr-users  On Behalf Of
>> >> > Maryam Baghdadi
>> >> > Sent: Tuesday, July 9, 2019 10:07 AM
>> >> > To: Kamailio (SER) - Users Mailing List
>> >> > 
>> >> > Subject: [SR-Users] kamailio as ICSCF connection with FHoss
>> >> >
>> >> >
>> >> >
>> >> > hello,
>> >> >
>> >> >
>> >> >
>> >> > I use Kamailio as my Icscf in IMS network .
>> >> >
>> >> > after receiving Invite Request from SCSCF to ICSCF‌, a LIR request
>> sent from ICSCF to FHoss. User_Unknown response received in all states. I
>> checked every thing carefully , users are registered and  defined in HSS
>> and every configuration done . What's wrong? is it possible that icscf has
>> problem? or it can be FHoss bug?
>>
>> --
> Henning Westerholt - https://skalatan.de/blog/
> Kamailio services - https://skalatan.de/services
>
>

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Re: [SR-Users] kamailio as ICSCF connection with FHoss

2019-07-14 Thread Mojtaba
Hello Henning,
According the 3GPP standards, the IMPU template would be started with '+',
for this reason in icscf, we have to add '+' to $rU, But in FHOSS the is
not any IMPU with '+'  prefix as default. There is two way to solve this
issue:
1) Add some new IMPU/IMPI according the 3GPP standards. (Be notice that
IMPU would be started with '+' or ' tel).
2) Have some changes in Kamailio routing block to escape some rules.
The is no needed to have some changes in ims_icscf module.

With Best Regards.Mojtaba

On Sun, Jul 14, 2019 at 12:38 AM Henning Westerholt  wrote:

> Hello Mojtaba,
>
> Let us wait for more details from the reporter before coming to a
> conclusion. Without complete knowledge about the scenario there is always
> some guessing involved.
>
> Cheers,
>
> Henning
>
> --
> Henning Westerholt - https://skalatan.de/blog/
> Kamailio services - https://skalatan.de/services
>
> -Original Message-
> From: sr-users  On Behalf Of Mojtaba
> Sent: Saturday, July 13, 2019 8:36 PM
> To: Kamailio (SER) - Users Mailing List ;
> maryam.baghd...@gmail.com
> Subject: Re: [SR-Users] kamailio as ICSCF connection with FHoss
>
> Hello Mariam,
> It is worse way that you made choice. There is no needed to change any
> things in source code.
> Just dive to kamailio.cfg file and remove "+" prefix  from $rU (IMPU) . It
> is very straightforward.
> WIth Best Regards. Mojtaba
>
> On Sat, Jul 13, 2019 at 2:55 PM Maryam Baghdadi 
> wrote:
> >
> > Thank you for your help,
> > I resolved my problem , there was a '+' in public_identity that isn't
> shown in pcap file and just by printing value public_identity in
> I_perform_location_information_request() recognized .
> > I replaced  public_identity = cscf_get_public_identity_from_requri(msg)
> with  cscf_get_to_uri(msg, _identity) then recompiled ims_icscf
> module and my problem resolved .
> >
> >
> >
> > On Wed, Jul 10, 2019 at 3:07 PM Mojtaba  wrote:
> >>
> >> Hello,
> >> of course not, Are you sure the IMPI/IMPU in LRI/LRA message there
> >> are exist exactly in HSS?
> >> Or may they are changed with some prefixes like "tel uri" or "+"
> >> With Regards. Mojtaba
> >>
> >> On Wed, Jul 10, 2019 at 2:05 PM Henning Westerholt 
> wrote:
> >> >
> >> > Hello Maryam,
> >> >
> >> >
> >> >
> >> > I am not an expert on the FHoss, but just some ideas how to
> investigate it further:
> >> >
> >> >
> >> >
> >> > Do you get some error message in FHoss?
> >> > Can you test with some other tool (maybe with a test script) that the
> FHoss works ok generally?
> >> > Any error message from ICSCF module in the Kamailio logging?
> >> >
> >> >
> >> >
> >> > Cheers,
> >> >
> >> >
> >> >
> >> > Henning
> >> >
> >> >
> >> >
> >> > --
> >> >
> >> > Henning Westerholt - https://skalatan.de/blog/
> >> >
> >> > Kamailio services - https://skalatan.de/services
> >> >
> >> >
> >> >
> >> > From: sr-users  On Behalf Of
> >> > Maryam Baghdadi
> >> > Sent: Tuesday, July 9, 2019 10:07 AM
> >> > To: Kamailio (SER) - Users Mailing List
> >> > 
> >> > Subject: [SR-Users] kamailio as ICSCF connection with FHoss
> >> >
> >> >
> >> >
> >> > hello,
> >> >
> >> >
> >> >
> >> > I use Kamailio as my Icscf in IMS network .
> >> >
> >> > after receiving Invite Request from SCSCF to ICSCF‌, a LIR request
> sent from ICSCF to FHoss. User_Unknown response received in all states. I
> checked every thing carefully , users are registered and  defined in HSS
> and every configuration done . What's wrong? is it possible that icscf has
> problem? or it can be FHoss bug?
> >> >
> >> >
> >> >
> >> > ___
> >> > 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
>
> ___
> 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] kamailio as ICSCF connection with FHoss

2019-07-14 Thread Mojtaba
Yes, In the request that you make, there isn't any '+', but when the
request come to icscf, the '+' is added to request. After that the LIR
message is issued.
If you have look on kamailio.cfg in icscf node, the code is appear:
# -  request routing logic ---
# main routing logic

route{
*.*
*.*
*.*


*if !($rU =~ "\+.*") {prefix("+");*
*}*
*.*
*.*
*.*
The '+' is added in above code. You just comment those codes to prevent
adding '+' in $rU.
With Best Regards.Mojtaba


On Sat, Jul 13, 2019 at 11:30 PM Mary Honey  wrote:

> No ,
> There isn't any  '+' in request msg . I know what I did
> In the function and after extracting public_identity one of the functions
> unexpectly adds '+' to public_identity
>
>
> On Sat, Jul 13, 2019, 11:08 PM Mojtaba 
>> Hello Henning,
>> I don't think so there is no needed to change any code in ims_icscf
>> module.
>> With Best Regards. Mojtaba
>>
>> On Sat, Jul 13, 2019 at 3:06 PM Henning Westerholt 
>> wrote:
>> >
>> > Hello Maryam,
>> >
>> > great that you were able to solve it. You mentioned that you need to
>> re-compile the ims_icscf module. Is there in your opinion an error or
>> limitation in this module?
>> >
>> > Cheers,
>> >
>> > Henning
>> >
>> > Am 13.07.19 um 12:24 schrieb Maryam Baghdadi:
>> >
>> > Thank you for your help,
>> > I resolved my problem , there was a '+' in public_identity that isn't
>> shown in pcap file and just by printing value public_identity in
>> I_perform_location_information_request() recognized .
>> > I replaced  public_identity = cscf_get_public_identity_from_requri(msg)
>> with  cscf_get_to_uri(msg, _identity) then recompiled ims_icscf
>> module and my problem resolved .
>> >
>> >
>> >
>> > On Wed, Jul 10, 2019 at 3:07 PM Mojtaba  wrote:
>> >>
>> >> Hello,
>> >> of course not, Are you sure the IMPI/IMPU in LRI/LRA message there are
>> >> exist exactly in HSS?
>> >> Or may they are changed with some prefixes like "tel uri" or "+"
>> >> With Regards. Mojtaba
>> >>
>> >> On Wed, Jul 10, 2019 at 2:05 PM Henning Westerholt 
>> wrote:
>> >> >
>> >> > Hello Maryam,
>> >> >
>> >> >
>> >> >
>> >> > I am not an expert on the FHoss, but just some ideas how to
>> investigate it further:
>> >> >
>> >> >
>> >> >
>> >> > Do you get some error message in FHoss?
>> >> > Can you test with some other tool (maybe with a test script) that
>> the FHoss works ok generally?
>> >> > Any error message from ICSCF module in the Kamailio logging?
>> >> >
>> >> >
>> >> >
>> >> > Cheers,
>> >> >
>> >> >
>> >> >
>> >> > Henning
>> >> >
>> >> >
>> >> >
>> >> > --
>> >> >
>> >> > Henning Westerholt - https://skalatan.de/blog/
>> >> >
>> >> > Kamailio services - https://skalatan.de/services
>> >> >
>> >> >
>> >> >
>> >> > From: sr-users  On Behalf Of
>> Maryam Baghdadi
>> >> > Sent: Tuesday, July 9, 2019 10:07 AM
>> >> > To: Kamailio (SER) - Users Mailing List > >
>> >> > Subject: [SR-Users] kamailio as ICSCF connection with FHoss
>> >> >
>> >> >
>> >> >
>> >> > hello,
>> >> >
>> >> >
>> >> >
>> >> > I use Kamailio as my Icscf in IMS network .
>> >> >
>> >> > after receiving Invite Request from SCSCF to ICSCF‌, a LIR request
>> sent from ICSCF to FHoss. User_Unknown response received in all states. I
>> checked every thing carefully , users are registered and  defined in HSS
>> and every configuration done . What's wrong? is it possible that icscf has
>> problem? or it can be FHoss bug?
>> >> >
>> >> >
>> >> >
>> >> > ___
>> >> > Kamailio (SER) - Users Mailing List
>> >> > sr-users@lists.kamailio.org
>> >> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>> >>
>> >>
>> >>
>> >> --
>> >> --Mojtaba 

Re: [SR-Users] kamailio as ICSCF connection with FHoss

2019-07-13 Thread Mojtaba
Hello Henning,
I don't think so there is no needed to change any code in ims_icscf module.
With Best Regards. Mojtaba

On Sat, Jul 13, 2019 at 3:06 PM Henning Westerholt  wrote:
>
> Hello Maryam,
>
> great that you were able to solve it. You mentioned that you need to 
> re-compile the ims_icscf module. Is there in your opinion an error or 
> limitation in this module?
>
> Cheers,
>
> Henning
>
> Am 13.07.19 um 12:24 schrieb Maryam Baghdadi:
>
> Thank you for your help,
> I resolved my problem , there was a '+' in public_identity that isn't shown 
> in pcap file and just by printing value public_identity in 
> I_perform_location_information_request() recognized .
> I replaced  public_identity = cscf_get_public_identity_from_requri(msg) with  
> cscf_get_to_uri(msg, _identity) then recompiled ims_icscf module and 
> my problem resolved .
>
>
>
> On Wed, Jul 10, 2019 at 3:07 PM Mojtaba  wrote:
>>
>> Hello,
>> of course not, Are you sure the IMPI/IMPU in LRI/LRA message there are
>> exist exactly in HSS?
>> Or may they are changed with some prefixes like "tel uri" or "+"
>> With Regards. Mojtaba
>>
>> On Wed, Jul 10, 2019 at 2:05 PM Henning Westerholt  wrote:
>> >
>> > Hello Maryam,
>> >
>> >
>> >
>> > I am not an expert on the FHoss, but just some ideas how to investigate it 
>> > further:
>> >
>> >
>> >
>> > Do you get some error message in FHoss?
>> > Can you test with some other tool (maybe with a test script) that the 
>> > FHoss works ok generally?
>> > Any error message from ICSCF module in the Kamailio logging?
>> >
>> >
>> >
>> > Cheers,
>> >
>> >
>> >
>> > Henning
>> >
>> >
>> >
>> > --
>> >
>> > Henning Westerholt - https://skalatan.de/blog/
>> >
>> > Kamailio services - https://skalatan.de/services
>> >
>> >
>> >
>> > From: sr-users  On Behalf Of Maryam 
>> > Baghdadi
>> > Sent: Tuesday, July 9, 2019 10:07 AM
>> > To: Kamailio (SER) - Users Mailing List 
>> > Subject: [SR-Users] kamailio as ICSCF connection with FHoss
>> >
>> >
>> >
>> > hello,
>> >
>> >
>> >
>> > I use Kamailio as my Icscf in IMS network .
>> >
>> > after receiving Invite Request from SCSCF to ICSCF‌, a LIR request sent 
>> > from ICSCF to FHoss. User_Unknown response received in all states. I 
>> > checked every thing carefully , users are registered and  defined in HSS 
>> > and every configuration done . What's wrong? is it possible that icscf has 
>> > problem? or it can be FHoss bug?
>> >
>> >
>> >
>> > ___
>> > 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
>
> --
> Henning Westerholt - https://skalatan.de/blog/
> Kamailio services - https://skalatan.de/services
>
> ___
> 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] kamailio as ICSCF connection with FHoss

2019-07-13 Thread Mojtaba
Hello Mariam,
It is worse way that you made choice. There is no needed to change any
things in source code.
Just dive to kamailio.cfg file and remove "+" prefix  from $rU (IMPU)
. It is very straightforward.
WIth Best Regards. Mojtaba

On Sat, Jul 13, 2019 at 2:55 PM Maryam Baghdadi
 wrote:
>
> Thank you for your help,
> I resolved my problem , there was a '+' in public_identity that isn't shown 
> in pcap file and just by printing value public_identity in 
> I_perform_location_information_request() recognized .
> I replaced  public_identity = cscf_get_public_identity_from_requri(msg) with  
> cscf_get_to_uri(msg, _identity) then recompiled ims_icscf module and 
> my problem resolved .
>
>
>
> On Wed, Jul 10, 2019 at 3:07 PM Mojtaba  wrote:
>>
>> Hello,
>> of course not, Are you sure the IMPI/IMPU in LRI/LRA message there are
>> exist exactly in HSS?
>> Or may they are changed with some prefixes like "tel uri" or "+"
>> With Regards. Mojtaba
>>
>> On Wed, Jul 10, 2019 at 2:05 PM Henning Westerholt  wrote:
>> >
>> > Hello Maryam,
>> >
>> >
>> >
>> > I am not an expert on the FHoss, but just some ideas how to investigate it 
>> > further:
>> >
>> >
>> >
>> > Do you get some error message in FHoss?
>> > Can you test with some other tool (maybe with a test script) that the 
>> > FHoss works ok generally?
>> > Any error message from ICSCF module in the Kamailio logging?
>> >
>> >
>> >
>> > Cheers,
>> >
>> >
>> >
>> > Henning
>> >
>> >
>> >
>> > --
>> >
>> > Henning Westerholt - https://skalatan.de/blog/
>> >
>> > Kamailio services - https://skalatan.de/services
>> >
>> >
>> >
>> > From: sr-users  On Behalf Of Maryam 
>> > Baghdadi
>> > Sent: Tuesday, July 9, 2019 10:07 AM
>> > To: Kamailio (SER) - Users Mailing List 
>> > Subject: [SR-Users] kamailio as ICSCF connection with FHoss
>> >
>> >
>> >
>> > hello,
>> >
>> >
>> >
>> > I use Kamailio as my Icscf in IMS network .
>> >
>> > after receiving Invite Request from SCSCF to ICSCF‌, a LIR request sent 
>> > from ICSCF to FHoss. User_Unknown response received in all states. I 
>> > checked every thing carefully , users are registered and  defined in HSS 
>> > and every configuration done . What's wrong? is it possible that icscf has 
>> > problem? or it can be FHoss bug?
>> >
>> >
>> >
>> > ___
>> > 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



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Re: [SR-Users] Kamailio Asterisk integration with docker DNS Problem

2019-07-13 Thread Mojtaba
Hello,
Did you add SRV record for each server in your DNS server?

On Sat, Jul 13, 2019 at 6:07 PM Fred Posner  wrote:
>
> The error indicates kamailio can not resolve the location of asterisk2. Try 
> checking the dns (can you resolve it on the kamailio box?) or using the IP.
>
> -- Fred
>
>
> On Jul 13, 2019, at 07:09, Amir  wrote:
>
> Hi
> I have created a kamailio container with docker and two asterisk container
> And this is my dispatcher list:
> 1 sip:asterisk:5060
> 1 sip:asterisk2:5060
> and this is my both asterisk SIP.conf
> [Kamailio]
> host=kamailio
> port=5060
> insecure=invite
> type=friend
> context=from-internal
>
> The problem is that I use User-defined networks which has its own DNS
> when a container starts a dns record would be set
> If I start a kamailio first then start two asterisks the asterisk works fine
> because the name "kamailio" has been set in docker dns
> but I should restart the kamailio because at startup It could not resolve 
> astersik dns records
> after the everything works fine but if the any of the asterisk instances 
> crash/stop kamailio
> can not detect that because a dns record would not be available:
>
> 17(22) ERROR:  [core/resolve.c:1684]: sip_hostport2su(): could not 
> resolve hostname: "asterisk2"
> 17(22) ERROR: tm [ut.h:309]: uri2dst2(): failed to resolve "asterisk2"
> 17(22) ERROR: tm [uac.c:452]: t_uac_prepare(): no socket found
> 17(22) ERROR: dispatcher [dispatch.c:3110]: ds_ping_set(): unable to ping 
> [sip:asterisk2:5060]
>
>
> It is kind of loop , I think maybe I can fix asterisk problem with asterisk 
> realtime and
> set the sip.conf in database.
> anyone has any suggestion for this problem?
>
> ___
> 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



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Re: [SR-Users] kamailio as ICSCF connection with FHoss

2019-07-10 Thread Mojtaba
Hello,
of course not, Are you sure the IMPI/IMPU in LRI/LRA message there are
exist exactly in HSS?
Or may they are changed with some prefixes like "tel uri" or "+"
With Regards. Mojtaba

On Wed, Jul 10, 2019 at 2:05 PM Henning Westerholt  wrote:
>
> Hello Maryam,
>
>
>
> I am not an expert on the FHoss, but just some ideas how to investigate it 
> further:
>
>
>
> Do you get some error message in FHoss?
> Can you test with some other tool (maybe with a test script) that the FHoss 
> works ok generally?
> Any error message from ICSCF module in the Kamailio logging?
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> --
>
> Henning Westerholt - https://skalatan.de/blog/
>
> Kamailio services - https://skalatan.de/services
>
>
>
> From: sr-users  On Behalf Of Maryam 
> Baghdadi
> Sent: Tuesday, July 9, 2019 10:07 AM
> To: Kamailio (SER) - Users Mailing List 
> Subject: [SR-Users] kamailio as ICSCF connection with FHoss
>
>
>
> hello,
>
>
>
> I use Kamailio as my Icscf in IMS network .
>
> after receiving Invite Request from SCSCF to ICSCF‌, a LIR request sent from 
> ICSCF to FHoss. User_Unknown response received in all states. I checked every 
> thing carefully , users are registered and  defined in HSS and every 
> configuration done . What's wrong? is it possible that icscf has problem? or 
> it can be FHoss bug?
>
>
>
> ___
> 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] ATCF/ATGW functionality

2019-07-01 Thread Mojtaba
According to ETSI TS 129-280 V15, Your network must support Sv
interface between MME to MSC.  Your network infrastructure (EPC)
should support this feature. So it is not intuitively related to PCSCF
or SBC.
Add external MSC server for what? Each 2g/3g network have it's MSC and
it would be support Sv interface, too.
With Best Regards.Mojtaba

On Sun, Jun 30, 2019 at 4:32 PM Maryam Baghdadi
 wrote:
>
> Thank you for your response,
> If we need to handover of call in our network, What's the solution? is it 
> possible to add external MSC Server to out network? or add a SBC with this 
> ability ?
>
> On Sun, Jun 30, 2019 at 3:40 PM Mojtaba  wrote:
>>
>> Unfortunately you can't, because SRVCC is a capable between MME and
>> MCS in your network. When the UE is moving on,  the MME send "Take
>> handover of  the call" to MSC and move the surrect session control to
>> MSC. You could read more about SRVCC in 3GPP TS 29.280.
>> And FHOSS does not support interface between MME and HSS,
>> With Best Regards.Mojtaba
>>
>> On Sun, Jun 30, 2019 at 11:28 AM Maryam Baghdadi
>>  wrote:
>> >
>> > Hello,
>> >
>> > Is it possible to add ATCF/ATGW functionality in kamailio  as P-cscf ?
>> > I have IMS network with kamailio as P-cscf, icscf, scscf and FHoss .
>> > ATCF/ATGW is necessary for handovers between 2G/3G and LTE networks.
>> >
>> > ___
>> > 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



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Re: [SR-Users] ATCF/ATGW functionality

2019-06-30 Thread Mojtaba
Unfortunately you can't, because SRVCC is a capable between MME and
MCS in your network. When the UE is moving on,  the MME send "Take
handover of  the call" to MSC and move the surrect session control to
MSC. You could read more about SRVCC in 3GPP TS 29.280.
And FHOSS does not support interface between MME and HSS,
With Best Regards.Mojtaba

On Sun, Jun 30, 2019 at 11:28 AM Maryam Baghdadi
 wrote:
>
> Hello,
>
> Is it possible to add ATCF/ATGW functionality in kamailio  as P-cscf ?
> I have IMS network with kamailio as P-cscf, icscf, scscf and FHoss .
> ATCF/ATGW is necessary for handovers between 2G/3G and LTE networks.
>
> ___
> 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?
> >>>>>

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

2019-06-12 Thread Mojtaba
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 >
>> 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 >>
>>> 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?
>>> >>
>>> >> Cheers,
>>> >> Daniel
>>> >>
>>> >> On 11.06.19 12:01, Mojtaba wrote:
>>> >>> Hello,
>>> >>> What does this CRITICAL error mean?
>>> >>> Jun 11 05:48:13 debian8 /usr/local/sbin/kamailio[8468]: {1 1 INVITE
>>> >>> Njg0MzNhMzJhOTY3MzQ3NmNlNWY0ODkzMTJmNTM1ZTk.} CRITICAL: 
>>> >>> [core/mem/q_malloc.c:486]: qm_free(): BUG: bad pointer 0xa3c188 (out
>>> >>> of memory block!) called from core: core/data_lump.c: free_lump(466) -
>>> >>> aborting
>>> >>>
>>> >>> I got it when i make new call during debugging new module in Kamailio?
>>> >>> In module we have to change some message's body with data_lump.
>>> >>> I asked relaed issue (strange behaviour while developing new module)
>>> >>> in Kamailio developer forum 
>>> >>> With Best Regards.
>>> >>>
>>> >> --
>>> >> Daniel-Constantin Mierla -- www.asipto.com
>>> >> www.twitter.com/miconda -- www.linkedin.com/in/miconda
>>> >>
>>> >
>>> > --
>>> > --Mojtaba Esfandiari.S
>>>
>>> --
>>> Daniel-Constantin Mierla -- www.asipto.com
>>> www.twitter.com/miconda -- www.linkedin.com/in/miconda
>>>
>> --
>> Daniel-Constantin Mierla -- www.asipto.com
>> www.twitter.com/miconda -- www.linkedin.com/in/miconda
>
> --
> Daniel-Constantin Mierla -- www.asipto.com
> www.twitter.com/miconda -- www.linkedin.com/in/miconda



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

2019-06-11 Thread Mojtaba
Thank you.
And what about the crash when using it with htable api module?

On Tue, 11 Jun 2019, 18:18 Daniel-Constantin Mierla  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?
>> >>
>> >> Cheers,
>> >> Daniel
>> >>
>> >> On 11.06.19 12:01, Mojtaba wrote:
>> >>> Hello,
>> >>> What does this CRITICAL error mean?
>> >>> Jun 11 05:48:13 debian8 /usr/local/sbin/kamailio[8468]: {1 1 INVITE
>> >>> Njg0MzNhMzJhOTY3MzQ3NmNlNWY0ODkzMTJmNTM1ZTk.} CRITICAL: 
>> >>> [core/mem/q_malloc.c:486]: qm_free(): BUG: bad pointer 0xa3c188 (out
>> >>> of memory block!) called from core: core/data_lump.c: free_lump(466) -
>> >>> aborting
>> >>>
>> >>> I got it when i make new call during debugging new module in Kamailio?
>> >>> In module we have to change some message's body with data_lump.
>> >>> I asked relaed issue (strange behaviour while developing new module)
>> >>> in Kamailio developer forum 
>> >>> With Best Regards.
>> >>>
>> >> --
>> >> Daniel-Constantin Mierla -- www.asipto.com
>> >> www.twitter.com/miconda -- www.linkedin.com/in/miconda
>> >>
>> >
>> > --
>> > --Mojtaba Esfandiari.S
>>
>> --
>> Daniel-Constantin Mierla -- www.asipto.com
>> www.twitter.com/miconda -- www.linkedin.com/in/miconda
>>
>> --
> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 
> www.linkedin.com/in/miconda
>
>
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Re: [SR-Users] CRITICAL ERROR in data_lamp.c!

2019-06-11 Thread Mojtaba
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  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?
> >>
> >> Cheers,
> >> Daniel
> >>
> >> On 11.06.19 12:01, Mojtaba wrote:
> >>> Hello,
> >>> What does this CRITICAL error mean?
> >>> Jun 11 05:48:13 debian8 /usr/local/sbin/kamailio[8468]: {1 1 INVITE
> >>> Njg0MzNhMzJhOTY3MzQ3NmNlNWY0ODkzMTJmNTM1ZTk.} CRITICAL: 
> >>> [core/mem/q_malloc.c:486]: qm_free(): BUG: bad pointer 0xa3c188 (out
> >>> of memory block!) called from core: core/data_lump.c: free_lump(466) -
> >>> aborting
> >>>
> >>> I got it when i make new call during debugging new module in Kamailio?
> >>> In module we have to change some message's body with data_lump.
> >>> I asked relaed issue (strange behaviour while developing new module)
> >>> in Kamailio developer forum 
> >>> With Best Regards.
> >>>
> >> --
> >> Daniel-Constantin Mierla -- www.asipto.com
> >> www.twitter.com/miconda -- www.linkedin.com/in/miconda
> >>
> >
> > --
> > --Mojtaba Esfandiari.S
>
> --
> Daniel-Constantin Mierla -- www.asipto.com
> www.twitter.com/miconda -- www.linkedin.com/in/miconda
>
>
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Re: [SR-Users] CRITICAL ERROR in data_lamp.c!

2019-06-11 Thread Mojtaba
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?
>
> Cheers,
> Daniel
>
> On 11.06.19 12:01, Mojtaba wrote:
> > Hello,
> > What does this CRITICAL error mean?
> > Jun 11 05:48:13 debian8 /usr/local/sbin/kamailio[8468]: {1 1 INVITE
> > Njg0MzNhMzJhOTY3MzQ3NmNlNWY0ODkzMTJmNTM1ZTk.} CRITICAL: 
> > [core/mem/q_malloc.c:486]: qm_free(): BUG: bad pointer 0xa3c188 (out
> > of memory block!) called from core: core/data_lump.c: free_lump(466) -
> > aborting
> >
> > I got it when i make new call during debugging new module in Kamailio?
> > In module we have to change some message's body with data_lump.
> > I asked relaed issue (strange behaviour while developing new module)
> > in Kamailio developer forum 
> > With Best Regards.
> >
> --
> Daniel-Constantin Mierla -- www.asipto.com
> www.twitter.com/miconda -- www.linkedin.com/in/miconda
>


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

2019-06-11 Thread Mojtaba
Hello,
What does this CRITICAL error mean?
Jun 11 05:48:13 debian8 /usr/local/sbin/kamailio[8468]: {1 1 INVITE
Njg0MzNhMzJhOTY3MzQ3NmNlNWY0ODkzMTJmNTM1ZTk.} CRITICAL: 
[core/mem/q_malloc.c:486]: qm_free(): BUG: bad pointer 0xa3c188 (out
of memory block!) called from core: core/data_lump.c: free_lump(466) -
aborting

I got it when i make new call during debugging new module in Kamailio?
In module we have to change some message's body with data_lump.
I asked relaed issue (strange behaviour while developing new module)
in Kamailio developer forum 
With Best Regards.

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Re: [SR-Users] run some extra things before relying message

2019-05-02 Thread Mojtaba
Hi.
I have tried the same things like it. I got sdp changed by rtpengine. But i
want to run some extra script before relying message.
It is strange the script will be run after the call stablished. It means
the script will be run at the same time the message is relying.

On Thu, 2 May 2019, 16:23 YASIN CANER,  wrote:

> Hello,
>
> "write_sdp_pv" parameter helps to write a pvalue that includes modified
> sdp. Could you try it?
>
> Yasin CANER
>
>
>
>
> https://kamailio.org/docs/modules/5.2.x/modules/rtpengine.html#rtpengine.p.write_sdp_pv
>
> ------
> *From:* sr-users  on behalf of
> Mojtaba 
> *Sent:* Thursday, May 2, 2019 2:06 PM
> *To:* Kamailio (SER) - Users Mailing List
> *Subject:* [SR-Users] run some extra things before relying message
>
> Hello,
> I have encountered with problem in Kamailio.
> At first glance it looks simple. Considering the Data Lump feature in
> Kamailio, All changes in SIP body, does not apply quickly.
> So if we are using RTPEngine or RTPProxy, How we can get body message
> (Actually IP and port from SDP after rtpengine_manage function) before
> relaying message?
> Suppose i want get some information and run some extra things before
> relying message?
> I just added my extra things in t_on_reply  block, but it's strange
> why it doesn't work before relaying!
>
> -Mojtaba Esfandiari.S
>
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[SR-Users] run some extra things before relying message

2019-05-02 Thread Mojtaba
Hello,
I have encountered with problem in Kamailio.
At first glance it looks simple. Considering the Data Lump feature in
Kamailio, All changes in SIP body, does not apply quickly.
So if we are using RTPEngine or RTPProxy, How we can get body message
(Actually IP and port from SDP after rtpengine_manage function) before
relaying message?
Suppose i want get some information and run some extra things before
relying message?
I just added my extra things in t_on_reply  block, but it's strange
why it doesn't work before relaying!

-Mojtaba Esfandiari.S

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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,
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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
> *>>* ___
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> *>>
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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] 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
> ___
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> sr-users@lists.kamailio.org
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>


-- 
--Mojtaba Esfandiari.S
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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] 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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>
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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
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
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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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] 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] 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
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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[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] 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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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

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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] 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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>>
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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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>
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>
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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] 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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>
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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] 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] 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] 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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