[SR-Users] Kamailio SIP Proxy solution

[Hai Bui Duc Ha]

Hi all,

I'm researching the solution on Kamilio like SIP forward message:


- At the moment, the customer has server IP PBX and their clients.
- Then, the client just point the SIP Porxy configure to Kamailio. After
that, Kamailio will forward to IP PBX and send the event to another server.
- Everything will be not changed for customer site.

I found the link do the similar thing but it still authenticate on
Kamailio:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb#kamailio_configuration

Do you have any advice ?

Regrads,
Hai Bui

-- 
Hai Bui
VoIP engineer, Cvoice team, HTK-HCM Office
Mobile: +84-165-618-9876
​
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[SR-Users] kamcmd uac.reg_info

[gh]

Hi all,

i am not able to execute kamcmd uac.reg_info and kamcmd uac.reg_enable.

http://kamailio.org/docs/modules/5.1.x/modules/uac.html#uac.r.uac.reg_info

kamcmd version
kamailio 5.0.1 (x86_64/linux)

Only the following uac related commands are available on my kamailio 
5.0.1 (x86_64/linux) box:


kamcmd ? |grep uac
tm.t_uac_start
tm.t_uac_wait

What's my mistake?

Best regards,
Georg

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Re: [SR-Users] Not able to replace route_uri in contact header uri

[Narayan P]

Hi Daniel,


I am attaching here my sip trace at client side.Request you to see this.

My client IP in register request is 125.16.231.74 and the server on which 
kamailio is running is 185.122.205.178.


I want my contact header to be 


I am using avp_subst() function to replace this.But kamailio is not starting.

Please reply if I am in wrong way or have different understanding as I am very 
new to kamailio.


Thanks,

Narayan




From: Narayan P
Sent: Friday, April 14, 2017 12:54:25 PM
To: sr-us...@lists.sip-router.org
Cc: moco...@gmail.com
Subject: RE: Not able to replace route_uri in contact header uri

Hi Daniel,

Thanks for your response.

I mean to say,
In my register request the route header i.e. route_uri is  
and contact header is @172.22.13.41.
I want my contact header to be @185.122.205.178.
So how can I replace the contact header's only domain part with route_uri.the 
user part of the contact header remains same.


Thanks,
Narayan

From: Narayan P
Sent: Thursday, April 13, 2017 11:13 AM
To: sr-us...@lists.sip-router.org
Subject: Re: Not able to replace route_uri in contact header uri

Hi ,

Can anybody help me how to replace the domain part of $route-uri in contact  
header .

I am using avp_subst().But kamailio is not starting.


I am new to kamailio.Any help will be highly appreciated.


Thanks,

Narayan


From: Narayan P
Sent: Wednesday, April 12, 2017 8:17:02 AM
To: sr-us...@lists.sip-router.org
Subject: Not able to replace route_uri in contact header uri


Hi All,

I am using pjsip client and kamailiio,both running on different servers.I want 
to replace the uri on which pjsip is running with the uri on which kamailio is 
running in contact header.
I put some logs on route[REGISTRAR] and able to see the $route_uri as kamailio 
server uri($avp(s:fs) and  $hdr(Contact) as pjsip server uri 
($avp(s:contact).But I am not able to replace the uri with avp_subst() function.

Can anybody help me how to replace the uri or with any any other function.

Below I have put the snippet of config file.


# Handle SIP registrations
route[REGISTRAR] {
if (!is_method("REGISTER")) return;

if(isflagset(FLT_NATS)) {
setbflag(FLB_NATB);
#!ifdef WITH_NATSIPPING
# do SIP NAT pinging
setbflag(FLB_NATSIPPING);
#!endif
}
xlog("Narayan: before sending contact\n");
$avp(s:contact) = $hdr(Contact);
$avp(s:fs) = $route_uri;
xlog("Narayan: Forced socket is $avp(s:fs)\n");
xlog("Narayan: contact header is $avp(s:contact)\n");
avp_subst("$avp(s:contact)/avp(s:contac)/g", "/(.*)@(.*)/$route_uri/");
#avp_pushto("$ru/domain","$fd");
#rewritehostport("185.122.206.62:5060");
t_on_reply("MANAGE_REPLY");
t_relay();
#if (!save("location")) {
#   sl_reply_error();
#}
exit;
}


Any suggestion will be highly appreciated.


Thanks,
Narayan


root@dharm-VirtualBox:/home/dharm/rudra/pjsip_comp/pjsip-apps/bin# 
root@dharm-VirtualBox:/home/dharm/rudra/pjsip_comp/pjsip-apps/bin# 
root@dharm-VirtualBox:/home/dharm/rudra/pjsip_comp/pjsip-apps/bin# 
root@dharm-VirtualBox:/home/dharm/rudra/pjsip_comp/pjsip-apps/bin# 
./pjsua-x86_64-unknown-linux-gnu --local-port=5067 
--outbound=sip:185.122.205.178:5070 --config-file=config_cfg.cfg --no-tcp
11:49:40.373 os_core_unix.c !pjlib 2.3 for POSIX initialized
11:49:40.373 sip_endpoint.c  .Creating endpoint instance...
11:49:40.373  pjlib  .select() I/O Queue created (0xe95240)
11:49:40.373 sip_endpoint.c  .Module "mod-msg-print" registered
11:49:40.373 sip_transport.  .Transport manager created.
11:49:40.373   pjsua_core.c  .PJSUA state changed: NULL --> CREATED
11:49:40.373   pjsua_core.c  .pjsua_init

11:49:40.373   pjsua_core.c  .rtpCompStatus 0 rtpEncStatus 0 sipCompressStatus 
1  encryptionState 1

11:49:40.373 sip_endpoint.c  .Module "mod-pjsua-log" registered
11:49:40.373 sip_endpoint.c  .Module "mod-tsx-layer" registered
11:49:40.373 sip_endpoint.c  .Module "mod-stateful-util" registered
11:49:40.373 sip_endpoint.c  .Module "mod-ua" registered
11:49:40.373 sip_endpoint.c  .Module "mod-100rel" registered
11:49:40.373 sip_endpoint.c  .Module "mod-pjsua" registered
11:49:40.373 sip_endpoint.c  .Module "mod-invite" registered
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
11:49:40.427   pa_dev.c  ..PortAudio sound library initialized, status=0
11:49:40.428   pa_dev.c  ..PortAudio host api count=2
11:49:40.428   pa_dev.c  ..Sound device count=11
11:49:40.428  pjlib  ..select() I/O Queue created (0xeddd98)
11:49:40.434