[SR-Users] Pause a Transaction for a short while

2016-05-05 Thread Sirvan Paraste
Hello all,

is there anyway to pause a transaction for a short while (1 second) ? I
have implemented the push notification, when the callee get register I need
a short time to be sure that the app is completely start and can do its
duty. otherwise it missed the call. so I want to pause the transaction.

I have tested async_sleep("1"); it worked. is there any better solution ?

$var(hjoin) = 0;
lock("$tU");
async_sleep("1");
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Re: [SR-Users] Pause a Transaction for a short while

2016-05-05 Thread Daniel-Constantin Mierla
Hello,

async_route() might be better, from the same module, given it makes it
clear that the context is switched to another routing block later. The
async_sleep() is also ok, just be aware of its side effects.

Cheers,
Daniel


On 05/05/16 10:03, Sirvan Paraste wrote:
>
> Hello all, 
>
> is there anyway to pause a transaction for a short while (1 second) ?
> I have implemented the push notification, when the callee get register
> I need a short time to be sure that the app is completely start and
> can do its duty. otherwise it missed the call. so I want to pause the
> transaction.
>
> I have tested async_sleep("1"); it worked. is there any better solution ?
>
> $var(hjoin) = 0;
> lock("$tU");
> async_sleep("1");
>
>
>
>
>
>
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-- 
Daniel-Constantin Mierla
http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, Berlin, May 18-20, 2016 - 
http://www.kamailioworld.com

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[SR-Users] Kamailio - Sip_capture Module

2016-05-05 Thread Ali Taher
Hello,

 

We are currently using kamailio sip_capture module, and then we need to
create a Wireshark file  (pcap format)from fields in the database.

As we noticed in sip_capture table there is msg fields that contains only
the sip content.

And in order to create our file we need to have the entire packet and not
part of it.

We are hopefully asking if you can give us a hand with that , beside the
Homer project, what we actually wanted is to know if there is another module
in kamailio that store the entire packet captured and not only the  part of
it.

 

Regards,

Ali Taher

 

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[SR-Users] Getting sip.instance in INVITE

2016-05-05 Thread Cibin Paul
Hi,

How can I  get sip.instance value in the sip header of INVITE? I do get this 
value during REGISTER

Thanks & Regards
Cibin



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Re: [SR-Users] Kamailio - Sip_capture Module

2016-05-05 Thread Alberto Sagredo
Someone answered your partner Zainab about a question quite similar to
yours.

Take a look

Thanks

2016-05-05 11:35 GMT+02:00 Ali Taher :

> Hello,
>
>
>
> We are currently using kamailio sip_capture module, and then we need to
> create a Wireshark file  (pcap format)from fields in the database.
>
> As we noticed in sip_capture table there is msg fields that contains only
> the sip content.
>
> And in order to create our file we need to have the entire packet and not
> part of it.
>
> We are hopefully asking if you can give us a hand with that , beside the
> Homer project, what we actually wanted is to know if there is another
> module in kamailio that store the entire packet captured and not only the
>  part of it.
>
>
>
> Regards,
>
> Ali Taher
>
>
>
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>
>
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Re: [SR-Users] Kamailio - Sip_capture Module

2016-05-05 Thread Ali Taher
Hi Alberto,

 

Thanks for your reply , but neither I nor my partner Zainab received the email 
you’re talking about.

Can you please share the reply with us.

 

Regards,

Ali

 

From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of 
Alberto Sagredo
Sent: Thursday, May 05, 2016 2:24 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Kamailio - Sip_capture Module

 

Someone answered your partner Zainab about a question quite similar to yours.

 

Take a look

 

Thanks

 

2016-05-05 11:35 GMT+02:00 Ali Taher mailto:ata...@vanrise.com> >:

Hello,

 

We are currently using kamailio sip_capture module, and then we need to create 
a Wireshark file  (pcap format)from fields in the database.

As we noticed in sip_capture table there is msg fields that contains only the 
sip content.

And in order to create our file we need to have the entire packet and not part 
of it.

We are hopefully asking if you can give us a hand with that , beside the Homer 
project, what we actually wanted is to know if there is another module in 
kamailio that store the entire packet captured and not only the  part of it.

 

Regards,

Ali Taher

 


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Re: [SR-Users] Kamailio - Sip_capture Module

2016-05-05 Thread Alberto Sagredo
Sure. Take a look to sip capture.org project. ITs what you need.

osé Seabra joseseab...@gmail.com a través de
 lists.sip-router.org
4 may. (hace 1 día)
para Kamailio
Hi Zeinab,

I think that this project (http://sipcapture.org/) already does what do you
need.

BR


2016-05-05 13:34 GMT+02:00 Ali Taher :

> Hi Alberto,
>
>
>
> Thanks for your reply , but neither I nor my partner Zainab received the
> email you’re talking about.
>
> Can you please share the reply with us.
>
>
>
> Regards,
>
> Ali
>
>
>
> *From:* sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On
> Behalf Of *Alberto Sagredo
> *Sent:* Thursday, May 05, 2016 2:24 PM
> *To:* Kamailio (SER) - Users Mailing List 
> *Subject:* Re: [SR-Users] Kamailio - Sip_capture Module
>
>
>
> Someone answered your partner Zainab about a question quite similar to
> yours.
>
>
>
> Take a look
>
>
>
> Thanks
>
>
>
> 2016-05-05 11:35 GMT+02:00 Ali Taher :
>
> Hello,
>
>
>
> We are currently using kamailio sip_capture module, and then we need to
> create a Wireshark file  (pcap format)from fields in the database.
>
> As we noticed in sip_capture table there is msg fields that contains only
> the sip content.
>
> And in order to create our file we need to have the entire packet and not
> part of it.
>
> We are hopefully asking if you can give us a hand with that , beside the
> Homer project, what we actually wanted is to know if there is another
> module in kamailio that store the entire packet captured and not only the
>  part of it.
>
>
>
> Regards,
>
> Ali Taher
>
>
>
>
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
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>
>
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[SR-Users] Third party registration help

2016-05-05 Thread Code Developer
Hello,
I'm after some help on third-party SIP registration in Kamailio - specifically, 
how to perform registration of user agents that don't register themselves, but 
only send out occasional messages. The UAs not registering themselves are 
devices on a GSM network that try to save band-width by not sending REGISTER 
requests for themselves. They only ever send out SIP messages when necessary, 
to which they expect a reply message from our processing server.
(Note, details like external IP addresses have been changed for security.)
The configuration is as follows. We have a SIP server running Kamailio (4.3, 
location db enabled, nathelper enabled) on a static external IP address 
(50.170.158.149), a processing server on an internal network registered with 
the SIP server (let's say with a User agent name of '1000'), and a monitoring 
device in another part of the country connected to the internet via a GSM 
network (let's say User agent name of '2000'). The device is effectively NAT'd 
on the GSM provider's private network.
At some point the device (UA 2000) sends a SIP message to the processing server 
via Kamailio (to sip:1000@50.170.158.149). The processing server is registered, 
so the SIP server knows where it is and forwards the message accordingly. 
200/OK replies propagate back successfully to the device.
The message flow appear as below. Only relevant header fields are shown:
Message received by the SIP server from remote device:

IP Src:60.250.149.198, Dst:50.170.158.149
UDP Src Port: 59177, Dst Port: 5060
--
Request-Line: MESSAGE sip:1000@50.170.158.149
Via: SIP/2.0/UDP 10.92.240.131:5060;branch=z9hG4bK...
To: 
From: ;tag=7a82c48a


Message forwarded from SIP server, received by processing server as shown:

IP Src:50.170.158.149, Dst: 192.168.0.235
UDP Src Port:5060, Dst Port:5060
--
Request-Line: MESSAGE sip:1000@50.170.158.146:5060; ob SIP/2.0
Via: SIP/2.0/UDP 50.170.158.149;branch=z9hG...
Via: SIP/2.0/UDP 
10.92.240.131:5060;rport=59177;received=60.250.149.198;branch=z9hG...
To: 
From: ;tag=7a82c48a


The processing server now needs to send back a reply message to the remote, 
unregistered device which is NATd on the GSM internal network. We can do this 
by sending back a reply to the external IP address and rport values found in 
the second Via header of the message received by the processing server 
(60.250.149.198, port 59177). The reply can only be sent from the SIP server or 
the GSM network's router will ignore it - it's only a temporary reply 
address/port open for a minute or so as it's effectively just a NAT route to 
the internet.
In the first instance, when the processing server attempts to send back a 
message to the remote device via the SIP server, the SIP server replies with a 
404 as the remote device hasn't registered and the SIP server has no idea where 
to send it.
After looking at RFC3261 my conclusion was that we must get the processing 
server to perform a third-party registration on behalf of the remote device 
before we can send a reply message back to that device.
My understanding is that to perform the third party registration I should send 
a REGISTER request from our processing server to the SIP server:
1) Set the 'From' field to the AOR of the UA performing the registration (which 
is the processing server, 1000).
2) Set the 'To' field to the AOR of the UA being registered (i.e. the remote 
device, 2000).
3) Add a 'Contact' header field set to the IP address and port of the remote 
device (from Via = 60.250.149.198 port 59177).

The message flow for the register is as follows (only relevant header fields 
shown):
Processing server sends a register request to the SIP Server:

Request-Line: REGISTER sip:50.170.158.149 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.235:5060;rport;branch=z9hG...
From: ;tag=e6e2d70...
To: 
Contact: 

SIP server responds to the processing server with a 401 unauthorized response 
containing the auth header field (nonce challenge):

WWW-Authenticate: Digest realm="50.170.158.149", 
nonce="VytkJ1crIvu4ekGp/M4mnRlYmJRciF3N"

Processing server sends a revised register request to the SIP Server, augmented 
with the authorisation header for the supplied nonce:

Request-Line: REGISTER sip:50.170.158.149 SIP/2.0
Via: SIP/2.0/UDP 50.170.158.146:1096; rport;branch=z9hG4...
From: ;tag=dae68a...
To: 
Contact: 
Authorization: Digest username="1000", realm="50.170.158.149", 
nonce="VytkJ1crIvu4ekGp/M4mnRlYmJRciF3N" response="d4d7ffba8..."

However, the SIP Server again responds with a 401 and won't perform the 
third-party registration. I haven't been able to determine why. I *think* the 
response hash the processing server is sending back is correct. I'm using PJSIP 
on our processing server to perform all SIP functionality. The processing 
server does obviously register itself, the difference being that the To and 
From fields are the same () and there is no Contact  
header.
I'm 

Re: [SR-Users] making rtpengine gives error on centos

2016-05-05 Thread Sergey Basov
Hi.

You can build rtpengine on another physical/virtual machine with
libevent2-devel installed or try to build "relocated' libevent2 rpm
with --prefix=/usr/local or build and install libevent2 by yourself
with --prefix=/usr/local...

devel packages you need only for building from source. You does not
need it on production system.
--
Best regards,
Sergey Basov e-mail: sergey.v.ba...@gmail.com

tel: (+38067) 403-62-54


2016-05-05 9:44 GMT+03:00 ycaner :
> Hello;
> libevent2 rpm was installed and it still gives error. i checked
> libevent_pthreads.so , it is installed.
>
> libevent2-devel conflicts with libevent-devel
>
> Thanks
>
>  ls /usr/lib64/libevent_p*
> /usr/lib64/libevent_pthreads-2.0.so.5
> /usr/lib64/libevent_pthreads-2.0.so.5.1.9
>
>
> rpm -Uvh libevent2-2.0.21-11.1.x86_64.rpm
> warning: libevent2-2.0.21-11.1.x86_64.rpm: Header V3 DSA/SHA1 Signature, key
> ID c86c6f35: NOKEY
> Preparing...###
> [100%]
>1:libevent2  ###
> [100%]
>
>
> rpm -Uvh libevent2-devel-2.0.21-11.1.x86_64.rpm
> warning: libevent2-devel-2.0.21-11.1.x86_64.rpm: Header V3 DSA/SHA1
> Signature, key ID c86c6f35: NOKEY
> Preparing...###
> [100%]
> file /usr/include/evdns.h from install of
> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from package
> libevent-headers-1.4.13-4.el6.noarch
> file /usr/include/event.h from install of
> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from package
> libevent-headers-1.4.13-4.el6.noarch
> file /usr/include/evhttp.h from install of
> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from package
> libevent-headers-1.4.13-4.el6.noarch
> file /usr/include/evrpc.h from install of
> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from package
> libevent-headers-1.4.13-4.el6.noarch
> file /usr/include/evutil.h from install of
> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from package
> libevent-headers-1.4.13-4.el6.noarch
> file /usr/bin/event_rpcgen.py from install of
> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from package
> libevent-devel-1.4.13-4.el6.x86_64
> file /usr/lib64/libevent.so from install of
> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from package
> libevent-devel-1.4.13-4.el6.x86_64
> file /usr/lib64/libevent_core.so from install of
> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from package
> libevent-devel-1.4.13-4.el6.x86_64
> file /usr/lib64/libevent_extra.so from install of
> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from package
> libevent-devel-1.4.13-4.el6.x86_64
> [root@host-185-88-7-208 ~]#
>
>
>
>
> --
> View this message in context: 
> http://sip-router.1086192.n5.nabble.com/making-rtpengine-gives-error-on-centos-tp148114p148139.html
> Sent from the Users mailing list archive at Nabble.com.
>
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Re: [SR-Users] making rtpengine gives error on centos

2016-05-05 Thread ycaner

  

  
  
Hello; 
      i removed libevent-devel and install libevent2-devel.kamailio
  doesnt affect it that i was thought about it 
      last time i installed rtpengine that there wasnt any problem
  about it so i thought it is bug. anyway 
  really thanks for help. 

2016-05-05 16:52 tarihinde Sergey Basov
  [via SIP Router] yazdı: 

 Hi.
  
  
  You can build rtpengine on another physical/virtual machine with
  
  libevent2-devel installed or try to build "relocated' libevent2
  rpm
  
  with --prefix=/usr/local or build and install libevent2 by
  yourself
  
  with --prefix=/usr/local...
  
  
  devel packages you need only for building from source. You does
  not
  
  need it on production system.
  
  --
  
  Best regards,
  
  Sergey Basov                     e-mail: [hidden email] 
  
  
  tel: (+38067) 403-62-54
  
  
  
  2016-05-05 9:44 GMT+03:00 ycaner < [hidden email] >:
  
> Hello;

> libevent2 rpm was installed and it still gives error. i
checked

> libevent_pthreads.so , it is installed.

>

> libevent2-devel conflicts with libevent-devel

>

> Thanks

>

>  ls /usr/lib64/libevent_p*

> /usr/lib64/libevent_pthreads-2.0.so.5

> /usr/lib64/libevent_pthreads-2.0.so.5.1.9

>

>

> rpm -Uvh libevent2-2.0.21-11.1.x86_64.rpm

> warning: libevent2-2.0.21-11.1.x86_64.rpm: Header V3
DSA/SHA1 Signature, key

> ID c86c6f35: NOKEY

> Preparing...              
 ###

> [100%]

>    1:libevent2            
 ###

> [100%]

>

>

> rpm -Uvh libevent2-devel-2.0.21-11.1.x86_64.rpm

> warning: libevent2-devel-2.0.21-11.1.x86_64.rpm: Header V3
DSA/SHA1

> Signature, key ID c86c6f35: NOKEY

> Preparing...              
 ###

> [100%]

>         file /usr/include/evdns.h from install of

> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from
package

> libevent-headers-1.4.13-4.el6.noarch

>         file /usr/include/event.h from install of

> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from
package

> libevent-headers-1.4.13-4.el6.noarch

>         file /usr/include/evhttp.h from install of

> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from
package

> libevent-headers-1.4.13-4.el6.noarch

>         file /usr/include/evrpc.h from install of

> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from
package

> libevent-headers-1.4.13-4.el6.noarch

>         file /usr/include/evutil.h from install of

> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from
package

> libevent-headers-1.4.13-4.el6.noarch

>         file /usr/bin/event_rpcgen.py from install of

> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from
package

> libevent-devel-1.4.13-4.el6.x86_64

>         file /usr/lib64/libevent.so from install of

> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from
package

> libevent-devel-1.4.13-4.el6.x86_64

>         file /usr/lib64/libevent_core.so from install of

> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from
package

> libevent-devel-1.4.13-4.el6.x86_64

>         file /usr/lib64/libevent_extra.so from install of

> libevent2-devel-2.0.21-11.1.x86_64 conflicts with file from
package

> libevent-devel-1.4.13-4.el6.x86_64

> [root@host-185-88-7-208 ~]#

>

>

>

>

> --

> View this message in context: 
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> Sent from the Users mailing list archive at Nabble.com.

>

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[SR-Users] User offline

2016-05-05 Thread 王欣鑫
Hello, 

I use kamailio set up an IMS on a cloud server in a public network, and have
alice and bob registered on the server using ImsDroid in a private network.

When I use alice to call bob, I got "User offline" error.

Below are .pcap file captured from 4060 port and
pcscf.log/icscf.log/scscf.log and all config files(in kamailio.zip)

4060.pcap   
pcscf.log   
icscf.log   
scscf.log   
kamailio.zip
  


My kamailio version and debian operating system version is like below:

root@iZ94lg22es7Z:~# kamailio -version
version: kamailio 4.4.0 (x86_64/linux) d4f23c
flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX,
FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR,
USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: d4f23c 
compiled on 13:02:46 May  5 2016 with gcc 4.9.2
root@iZ94lg22es7Z:~# 

root@iZ94lg22es7Z:~# cat /etc/issue
Debian GNU/Linux 8 \n \l

root@iZ94lg22es7Z:~# 


Can anybody give me some help?

Any suggestion will be appreciated!



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Re: [SR-Users] User offline

2016-05-05 Thread 王欣鑫
Is this a NAT issue?

The network is like this:

alice--192.168.199.214 company ip--116.228.89.198kamailio IMS(cloud
server)--120.76.118.128
bob--192.168.199.124

I can see INVITE was sent from 116.228.89.198(alice) to 
120.76.118.128(kamailio IMS), then 120.76.118.128(kamailio IMS) reply a 100
trying to 116.228.89.198(alice), after that 120.76.118.128(kamailio IMS)
sent INVITE to 192.168.199.124(bob)
However, I did not captured bob's 183/180 or 200 OK.

What should I do to let it run properly?

 



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Re: [SR-Users] User offline

2016-05-05 Thread 王欣鑫
Is this a NAT issue? 

The network is like this: 

alice--192.168.199.214 company ip--116.228.89.198kamailio IMS(cloud
server)--120.76.118.128 
bob--192.168.199.124 

I can see INVITE was sent from 116.228.89.198(company ip) to 
120.76.118.128(kamailio IMS), then 120.76.118.128(kamailio IMS) reply a 100
trying to 116.228.89.198(company ip), after that 120.76.118.128(kamailio
IMS) sent INVITE to 192.168.199.124(private ip of bob), because 
192.168.199.124 is a local ip address, I guess kamailio IMS in the public ip
address can't find a route to this address.
So, bob did not receive this INVITE message from alice.

Is my analysis right? What should I do to let the IMS run properly? 



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Re: [SR-Users] User offline

2016-05-05 Thread 王欣鑫
Is this a NAT issue? 

The network is like this: 

alice--192.168.199.214 company ip--116.228.89.198kamailio IMS(cloud
server)--120.76.118.128 
bob--192.168.199.124 

I can see INVITE was sent from 116.228.89.198(company ip) to 
120.76.118.128(kamailio IMS), then 120.76.118.128(kamailio IMS) reply a 100
trying to 116.228.89.198(company ip), after that 120.76.118.128(kamailio
IMS) sent INVITE to 192.168.199.124(private ip of bob), because 
192.168.199.124 is a local ip address, I guess kamailio IMS in the public ip
address can't find a route to this address. 
So, bob did not receive this INVITE message from alice. 

Is my analysis right? What should I do to let the IMS run properly?

 



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Re: [SR-Users] User offline

2016-05-05 Thread ycaner
hello;
try to open nat_travesal module on registrar server. your analyz is right in
my view. INVITE packets never reach to BOB.

cheers.



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Re: [SR-Users] User offline

2016-05-05 Thread 王欣鑫
Thanks for your reply!

I found nat_traversal.so in /usr/local/lib64/kamailio/modules/ directory.
But didn't find a switch in pcscf.cfg or kamailio.cfg. 

So how can I open nat_travesal module?



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Re: [SR-Users] User offline

2016-05-05 Thread 王欣鑫
Hello,ycaner!

Thanks for your advice!

I searched nat_traversal in kamailio.cfg in PCSCF module, and found that I
should define WITH_NATPING.

After doing this, a new problem came. What does this line mean? What should
I configure to replace
"sip:keepalive@HOSTNAME" ? Is this HOSTNAME a turn/stun server?

modparam("nat_traversal", "keepalive_from", *"sip:keepalive@HOSTNAME"*)


#!ifdef WITH_NATPING
# - nat_traversal params -
# If another keepalive is wanted, this is the place
modparam("nat_traversal", "keepalive_interval", 20)
# If another method than NOTIFY is wanted:
modparam("nat_traversal", "keepalive_method", "OPTIONS")
# From?
modparam("nat_traversal", "keepalive_from", "sip:keepalive@HOSTNAME")
# Where we store information about keep-alives:
modparam("nat_traversal", "keepalive_state_file",
"/var/run/kamailio/keepalive_state")
#!endif






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Re: [SR-Users] User offline

2016-05-05 Thread ycaner

  

  
  
Hello; 
  at registration route , you can use nat traversal function , there
  is a sample on kamailio.cfg 
  that parameter is just adding from and request header for
  information on OPTIONS packet. you can add your 120.XXX.XXX.XXX 
  
  Cheers. 
  
  

2016-05-06 08:59 tarihinde 王欣鑫 [via SIP
  Router] yazdı: 

 Hello,ycaner!
  
  
  Thanks for your advice!
  
  
  I searched nat_traversal in kamailio.cfg in PCSCF module, and
  found that I should define WITH_NATPING.
  
  
  After doing this, a new problem came. What does this line mean?
  What should I configure to replace
  
  "sip:keepalive@HOSTNAME" ? Is this HOSTNAME a turn/stun server?
  
  
  modparam("nat_traversal", "keepalive_from", "sip:keepalive@HOSTNAME")
  
  
  
  #!ifdef WITH_NATPING
  
  # - nat_traversal params -
  
  # If another keepalive is wanted, this is the place
  
  modparam("nat_traversal", "keepalive_interval", 20)
  
  # If another method than NOTIFY is wanted:
  
  modparam("nat_traversal", "keepalive_method", "OPTIONS")
  
  # From?
  
  modparam("nat_traversal", "keepalive_from",
  "sip:keepalive@HOSTNAME")
  
  # Where we store information about keep-alives:
  
  modparam("nat_traversal", "keepalive_state_file",
  "/var/run/kamailio/keepalive_state")
  
  #!endif
  
  
  
  
  
  
  
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Re: [SR-Users] User offline

2016-05-05 Thread 王欣鑫
Hello ycaner!

I did as you told me and finally it worked!

Here are my finall config files.

hss_config.zip
  
kamailio.zip
  

Thank you very much!!!



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Re: [SR-Users] User offline

2016-05-05 Thread 王欣鑫
Hello,ycaner!

Now I use a real IMS user data to register this user(for example
18616278715) to China Unicom's IMS through kamailio.

I can register successsfully, and can make call using this number, the
callee can pick up phone. However
neither party can hear counter part.

So, this should be rtp problem. And I captured a .pcap file on port 5060
like below. In this, I see in SDP information
it says 192.168.199.124, it's a  private address. So maybe the callee can't
setup a circuit to this ip, and thus no media was transfered? 

Should I uncomment RTPPROXY_ADDRESS, as in my pcscf.cfg there are two lines:

# IP-Adress(es) of the RTP-Proxy 
##!define RTPPROXY_ADDRESS "udp:10.1.2.186:2 udp:10.1.27.217:2"
##!define RTPPROXY_ADDRESS "udp:10.1.2.186:2"

This is the .pcap file captured on port 5060.
5060.pcap   

 






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