hi
i am transferring tls protocol to udp protocol.
client <---tls----> kamilio <----udp---> asterisk
how can i add in contact "transport=tls" when i am sending 200 OK
message.
please help me.
any suggestion is appreciated.
Hi List,
I am running Debian lenny. Can I just change the repos in sources.list from lenny to testing and unstable to get the correct python dependencies etc for the ag-project tools?
Thanks,
Brian
Hello,
Our VoIP provider doesn't provide CNAM to us so we have to access an
outside provider for this. Can anyone point me in the right direction
for running an http request and passing the result back out so the pbx
can pick grab the callerid. I know that OpenSER they has the Utils
module but SER doesn't have that. Any ideas?
Thanks in advance,
Brian Artigas
Allstate Computers, LLC
brian(a)allstatecomputers.com
www.allstatecomputers.com
P: 561.743.1521
F: 561.575.9234
Hi,
Does anyone know how I can do remote management of Kamailio in a secure way?
Looking at the documentation for MI_DATAGRAM and MI_XMLRPC, there does not
seem to be a way to authenticate the request, or a way to restrict requests
to specific IP ranges. I want to be able to send management commands
remotely, but I don't to make this wide open.
Thanks,
Brian.
My rtpproxy works now basically i am using kamailio 1.2 with LCR, but when i
dial one number via LCR it dial this number and it works but when i dial
second number it says "*filtered destination*", i have to restart the
service and then it dial second number...
i think LCR not de-alocating the resources any idea ?
2009/11/12 Denis Putyato <denis7979(a)mail.ru>
> 1) I would done not ” modparam("nathelper", "rtpproxy_sock", "udp:
> 127.0.0.1:8899")” but “modparam("nathelper", "rtpproxy_sock",
> "unix:/var/run/rtpproxy.sock")”
>
> 2) http://www.kamailio.org/docs/modules/1.5.x/nathelper.html#id2468157
>
>
>
> *From:* toqeer ali [mailto:toqeer83@gmail.com]
> *Sent:* Thursday, November 12, 2009 2:10 PM
>
> *To:* Denis Putyato
> *Cc:* users(a)lists.kamailio.org
> *Subject:* Re: [Kamailio-Users] Register Request Forward
>
>
>
> Thanks Denis,
>
>
>
> It is indeed the problem of NAT. We have tried with live ip and its work
> fine. However, behind the NAT it is not working. I have installed RTPproxy ,
> but still it is not working. Here are the configuration that we use in our
> cfg file.
>
>
>
> modparam("nathelper","received_avp", "$avp(i:42)")
> modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:8899")
> modparam("nathelper", "natping_interval", 30)
> modparam("nathelper", "ping_nated_only", 0)
> modparam("nathelper", "sipping_bflag", 7)
> modparam("nathelper", "sipping_from", "sip:pinger@127.0.0.1<sip%3Apinger(a)127.0.0.1>
> ")
>
> route {
>
> ## NAT Detection
> #
> force_rport();
> if (nat_uac_test("19")) {
> if (method=="REGISTER") {
> fix_nated_register();
> } else {
> fix_nated_contact();
> };
> setflag(5);
> };
>
>
> .................
>
>
>
> and rtpproxy is also configured on the same server.
>
>
>
>
>
>
> 2009/11/12 Denis Putyato <denis7979(a)mail.ru>
>
> No two way communication is a problem of NAT in general. If session has
> been established then it’s not codec problem.
>
>
>
> *From:* toqeer ali [mailto:toqeer83@gmail.com]
> *Sent:* Thursday, November 12, 2009 9:25 AM
>
>
> *To:* Denis Putyato
> *Cc:* users(a)lists.kamailio.org
> *Subject:* Re: [Kamailio-Users] Register Request Forward
>
>
>
> i have a problem with codec. when i call to canada from kamailio to PSTN, i
> could not listen his/her voice as they listen( no two way communication)
>
> but when call to other countries it works... Basically call to canada
> support g729 can i define any codec information in kamailio to G729.
>
>
>
> Basically the problem is only with local number when you dial with
> extension 1 and when you dial with 08 it works... 08 is international
> gateway of canada...
>
>
>
> X-lite i am using as a softphone.
>
>
>
> please help
>
>
>
> 2009/11/11 Denis Putyato <denis7979(a)mail.ru>
>
> “can i use regular expression in $fu=XXX”
>
>
>
> You can use regexp. $fu=~”expression”
>
>
>
> *From:* toqeer ali [mailto:toqeer83@gmail.com]
> *Sent:* Wednesday, November 11, 2009 3:08 PM
>
>
> *To:* Denis Putyato
> *Cc:* users@lists.kamailio..org <users(a)lists.kamailio.org>
> *Subject:* Re: [Kamailio-Users] Register Request Forward
>
>
>
> Hmmm
>
>
>
> it works now i just remove Record Route function and it works,,, now i can
> dial from 602 to 601... thanks :).
>
>
>
> i will start work on how i will register multiple extension via kamailio to
> asterisk ... can i use regular expression in $fu=XXX instead of actual
> extension extension...
>
> 2009/11/11 Denis Putyato <denis7979(a)mail.ru>
>
> Registration is a process for telling SoftSwitch (in your variant -
> Asterisk) where (on which username, IP address and port) the SoftSwitch must
> send INVITE when received incoming call for 601. Try to understand where
> your asterisk try to send INVITE when you try to call from 602 to 601.
>
> ngrep utility will help you (http://ngrep.sourceforge.net/download.html)
>
>
>
> *From:* toqeer ali [mailto:toqeer83@gmail.com]
> *Sent:* Wednesday, November 11, 2009 2:18 PM
>
>
> *To:* Denis Putyato
> *Cc:* users@lists.kamailio..org <users(a)lists.kamailio.org>
> *Subject:* Re: [Kamailio-Users] Register Request Forward
>
>
>
> Here is the update:
>
>
>
> what i have done so for is that I have "Register extension 601" to asterisk
> via kamailio and it is registered!.
>
>
>
> When i call to 602 (configured on 192.168.0.1 -- asterisk box) it calls and
> make connection and voice call also works (RTP Traffic is ok).
>
>
>
> However, when i call to 601 from 602 (which is directly connected to
> asterisk without kamailio ), 601 can't receive call via kamailio. This is
> the main problem. Please guide me that how i will define this in kamailio
> (ip = 192.168.0.2) that xlite with extension 601 can receive call through
> 602 which is another extension on Asterisk.
>
>
>
> In a nutshell: 601 (registered through Kamailio to Asterisk) ---can call
> ---> 602 (registered via Asterisk)
>
> However: 602----cannot call ------> 601
>
>
>
> please help.
>
>
>
>
>
> 2009/11/11 Denis Putyato <denis7979(a)mail.ru>
>
> Try to use dlg_bridge(); function in dialog module
>
>
>
> *From:* toqeer ali [mailto:toqeer83@gmail.com]
> *Sent:* Wednesday, November 11, 2009 12:13 PM
>
>
> *To:* Denis Putyato
> *Cc:* users@lists.kamailio..org <users(a)lists.kamailio.org>
> *Subject:* Re: [Kamailio-Users] Register Request Forward
>
>
>
>
> "that client send INVITE to asterisk via kamailio but no replay is in on
> this INVITE"
>
>
>
> basically Asterisk receive the request and send back to kamailio but
> kamailio can't send back to xlite..
>
>
>
> 2009/11/11 Denis Putyato <denis7979(a)mail.ru>
>
> «but call does not route back to me»
>
>
>
> You mean, that incoming call from asterisk to client via kamailio doesn’t
> work or that client send INVITE to asterisk via kamailio but no replay is in
> on this INVITE?
>
>
>
> *From:* toqeer ali [mailto:toqeer83@gmail.com]
> *Sent:* Wednesday, November 11, 2009 10:16 AM
>
>
> *To:* Denis Putyato
> *Cc:* users@l
>
> ists.kamailio.org
>
>
> *Subject:* Re: [Kamailio-Users] Register Request Forward
>
>
>
> Thanks Again for the prompt reply... Basicaly i want to route Register
> Requests for multiple asterisk via kamailio ... by using pattern matching
> of extensions...
>
>
>
> is i am on the right path ?? or i can achieve this from dispatcher or load
> balancer module .... Basically i want to register specific extensions
> to specific asterisk boxes....
>
>
>
>
>
> from your previous reply i can send call to asterisk and register to as
> well but call does not route back to me... any clue ...
>
> 2009/11/11 Denis Putyato <denis7979(a)mail.r
>
> “The secret is that the other sip server was asterisk and i want route
> call to asterisk via kamailio :).”
>
> I don’t understand why do you want to do any session with asterisk via
> kamailio… not directly to asterisk… but any way the decision is same –
> send() or rewritehost()
>
>
>
> route {
>
> if ($fU==”…”&&is_method(“INVITE”))
>
> record_route();
>
> rewritehost(“192.168.0..1”);
>
> t_relay();
>
> }
>
>
>
> Do this before authentication checking is made in kamailio.
>
> *From:* toqeer ali [mailto:toqeer83@gmail.com]
> *Sent:* Wednesday, November 11, 2009 6:49 AM
> *To:* Denis Putyato
> *Cc:* users(a)lists.kamailio.org
> *Subject:* Re: [Kamailio-Users] Register Request Forward
>
>
>
> Thanks it Registered but i can't call on this .. "authentication Required
> from proxy". what i should do now.
>
>
>
> The secret is that the other sip server was asterisk and i want route call
> to asterisk via kamailio :).
>
> 2009/11/10 Denis Putyato <denis7979(a)mail.ru>
>
> Hi
>
> If not secret, why do you want to use such scheme?
>
>
>
> Your decision
>
> send(); - core function.
>
>
>
> {
>
> If (is_method(“REGISTER”)&&$fU==”….”) send(“udp:192.168.0.1:5060”);
>
> }
>
>
>
> or
>
>
>
> {
>
> If (is_method(“REGISTER”)&&$fU==”….”) {
>
> rewritehost(“192.168..0.1”);
>
> t_relay();
>
> }
>
> }
>
>
>
> *From:* users-bounces(a)lists.kamailio.org [mailto:
> users-bounces(a)lists.kamailio.org] *On Behalf Of *toqeer ali
> *Sent:* Tuesday, November 10, 2009 1:56 PM
> *To:* users(a)lists.kamailio.org
> *Subject:* [Kamailio-Users] Register Request Forward
>
>
>
> Hi all,
>
>
>
> My question is how to forward Register Request to another sip server.
>
>
>
> For example one user (X-lite) want to register with 192.168.0.1(sip
> server1) via 192.168.0.2(sip server 2).
>
> it means that in the domain part of xlite is 192.168...0.2 and
> it basically register with 192.168.0.1.
>
> 192.168.0.2 will forward Register request to 192.168...0.1.
>
> it will be very helpful if any one can answer.
>
>
>
> Thanks
>
>
> --
> Toqeer Ali Syed
>
> Red Hat Certified Engineer
> mob: +92 321 9059916
>
>
>
>
> --
> Toqeer Ali Syed
>
> Red Hat Certified Engineer
> mob: +92 321 9059916
>
>
>
>
> --
> Toqeer Ali Syed
>
> Red Hat Certified Engineer
> mob: +92 321 9059916
>
>
>
>
> --
> Toqeer Ali Syed
>
> Red Hat Certified Engineer
> mob: +92 321 9059916
>
>
>
>
> --
> Toqeer Ali Syed
>
> Red Hat Certified Engineer
> mob: +92 321 9059916
>
>
>
>
> --
> Toqeer Ali Syed
>
> Red Hat Certified Engineer
> mob: +92 321 9059916
>
>
>
>
> --
> Toqeer Ali Syed
>
> Red Hat Certified Engineer
> mob: +92 321 9059916
>
>
>
>
> --
> Toqeer Ali Syed
>
> Red Hat Certified Engineer
> mob: +92 321 9059916
>
--
Toqeer Ali Syed
Red Hat Certified Engineer
mob: +92 321 9059916
whether skype or voipstunt are using SIP protocols. I have experienced that
in many countries, where voice ports are blocked by the Telecom providers
but still voipstunt and skype works for calling to PSTN.
Please give me some links, for further reading on this.
Hi all,
My question is how to forward Register Request to another sip server.
For example one user (X-lite) want to register with 192.168.0.1(sip server1)
via 192.168.0.2(sip server 2).
it means that in the domain part of xlite is 192.168.0.2 and
it basically register with 192.168.0.1.
192.168.0.2 will forward Register request to 192.168.0.1.
it will be very helpful if any one can answer.
Thanks
--
Toqeer Ali Syed
Red Hat Certified Engineer
mob: +92 321 9059916
Let me describe my scenario:
SipUser1 -------register request------------> kamailio (SIP Proxy) -------
forwarded register request-----------> Asterisk
1. In the above scenario, when I register to Asterisk through Kamailio,
it works fine when I give the "record-route()" function in the cfg file.
Further, using this, I can also call to another extension 602 which is also
registered to Asterisk but 602 cannot call to 601 which is registered via
kamailio.
2. However, If I remove the "record-route()" function without
unregistering 601 and restarting the server, 602 can call to 601 .
route {
if ($fU=="601"&&is_method("INVITE"))
record_route();
rewritehost("10.0.10.111"); //asterisk host
route(1);
}
route[1] {
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
exit;
}
should i add anything in script further...
--
Toqeer Ali Syed
Red Hat Certified Engineer
mob: +92 321 9059916
Is it possible to have kamailio (1.4.4) to listen on more than one port? Is
it as simple as adding another ³port=xxxx² in the config?
Sven Schulz
Penn State University
Telecommunications and Network Services
814.865.6116
sip:sven@psu.edu
Hi,
I am with openser about my first 40 hours. So far i managed to install
3.0. Was able to make calls between registered phones. Enabled
accounting, etc. Looks i have control ;) Now i have a very simple
question. I want to send calls for land lines via voip provider,
sipdiscount.com in my case. I want to enable so registered client agent
can send invite to that server, but it should re authenticate somehow. I
was looking for proxy_authorize www_authorize but it looks like it
authenticate my phones only.
I was trying rewrite rewritehostport, but of course was getting
authentications errors. Is there any manual, previous discussion on
this topic?
Thank you.
