Re: [SR-Users] FW: Kamailio as sipRedirect
Hello,
this is a static string: ”sip:.*@192.168.111.138"
It means you redirect any number to '.*'.
To preserve the incoming number, use:
$ru = "sip:" + $rU + "@192.168.111.138";
Cheers,
Daniel
On 9/6/12 3:40 PM, Fatima
Chahrour~Vanrise Support wrote:
Dear Daniel,
Attached is my
config file.
Is it correct
to set $ru=”sip:.*@192.168.111.138";
in case I need any number to be redirected to
192.168.111.138.
Thanks.
From:
sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On
Behalf Of Fatima Chahrour~Vanrise Support
Sent: Wednesday, September 05, 2012 5:19 PM
To: mico...@gmail.com
Cc: 'SIP Router - Kamailio (OpenSER) and SIP
Express Router (SER) - Users Mailing List'
Subject: Re: [SR-Users] FW: Kamailio as
sipRedirect
Hi Daniel,
The call is not
redirected to 1st kamailio, below is the call
trace where 2nd kamailio(192.168.111.139) send
request to 192.168.111.15 which is the gateway in the lcr
rule instead of sending this invite to 192.168.111.138.
U 192.168.111.139:5060 -
192.168.111.15:5060 (shouldn’t
this be 192.168.111.138:5060 instead??)
INVITE sip:22#96135050@192.168.111.15
SIP/2.0.
Record-Route:
sip:192.168.111.139;lr=on.
Record-Route:
sip:192.168.111.138;lr=on.
Via: SIP/2.0/UDP
192.168.111.139;branch=z9hG4bK5757.5b3b1975.0.
Via: SIP/2.0/UDP
192.168.111.138;branch=z9hG4bK5757.50ff9264.0.
Via: SIP/2.0/UDP
192.168.52.139:12650;received=192.168.111.11;branch=z9hG4bK-d8754z-85008205ba3ea607-1---d8754z-;rport=12650.
Max-Forwards: 68.
Contact:
sip:106@192.168.111.11:12650.
To:
"96135050"sip:96135050@192.168.111.138.
From:
"106"sip:106@192.168.111.138;tag=56645478.
Call-ID:
ZjJlYWYzZmJmYmJlMGE0MmYxYWUwYjYwZmY5MmJkMjg..
CSeq: 1 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
Content-Type: application/sdp.
User-Agent: X-Lite release 1100l stamp
47546.
Content-Length: 477.
This is my
Kamailio routing logic configured on the 1st Kamailio to
forward any call from 1st kamialio to 2nd
kamailio:
if
(is_method("INVITE"))
{
Sethostport(“192.168.111.139:5060”);
Append_branch();
}
My problem is
hereà This is my Kamailio routing
configured on 2nd kamailio which I want from it
to send back the call, with lcr_gateway already defined, to
1st Kamailio:
if
(is_method("INVITE"))
{
if (!load_gws("1")) {
sl_send_reply("503", "Unable to load gateways");
exit;
}
if(!next_gw()){
sl_send_reply("503", "Unable to find a gateway");
exit;}
route(1);
$ru="sip:.*@192.168.111.138";
send_reply("302","rEdIrEcTeD");
km_append_branch("uri");
}
Am looking
through many documents but can’t achieve my target, I want
the 2nd Kamailio, which has LCR configured on it,
to send back to 1st Kamailio the gateway from
lcr_gateways with 302 redirect message.
My questions
are:
How to achieve
this?
Do you suggest
any document that help?
Do I need to
load any specific module?
Thanks,
F Chahrour
From:
Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Sent: Wednesday, September 05, 2012 9:38 AM
Re: [SR-Users] FW: Kamailio as sipRedirect
Hello,
On 9/6/12 6:47 PM, Fatima
Chahrour~Vanrise Support wrote:
Hi again,
I really need
your help in my testing lab here!!
I have
simplified the scenario and changed configs to make it easy
pinpoint the redirect issue and then apply on my real
scenario:
Two
Phones(username: 106 and 107) registered on Kamailio1
(192.168.111.138).
-
Phone 106: dial 107 àOK
-
Kamailio routing forward call to
Kamailio2(192.168.111.139) àOK
-
Then Kamailio should only receive the
call from Kamailio1 and redirect it back to Kamailio1 àNOT OK
Note: In my real scenario Kamailio 2
will be responsible for taking the routing decision based on
its defined LCR rules and redirect the call with rewritten
uri to Kamailio1.
Setting in
Kamailio2 routing script the following is not working till
now:
$ru = "sip:107@192.168.111.138";
send_reply("302", "Redirected");
the two lines above should be enough for the redirect. I wrote in
another email, to preserve the incoming username, use:
$ru = "sip:" + $rU + "@192.168.111.138";
If it is not working, set debug=3 and watch the log messages (in
/var/log/syslog or /var/log/messages) to understand what is not
configured properly.
Km_append_branch(“uri”);
This is not necessary and the parameter has to be a valid SIP URI --
you copied the prototype as it is in docs, defining the format of
the function.
Cheers,
Daniel
Kamailio2 is
not sending the 302 message and this is my problem.
How can I send
the 302 message redirect? And where do I have to place or
put the related function in the script?
Thank you.
Kind regards,
F.Chahrour
From:
sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On
Behalf Of Fatima Chahrour~Vanrise Support
Sent: Thursday, September 06, 2012 4:40 PM
To: 'SIP Router - Kamailio (OpenSER) and SIP
Express Router (SER) - Users Mailing List';
mico...@gmail.com
Subject: Re: [SR-Users] FW: Kamailio as
sipRedirect
Dear Daniel,
Attached is my
config file.
Is it correct
to set $ru=”sip:.*@192.168.111.138"; in case I need any
number to be redirected to 192.168.111.138.
Thanks.
From:
sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On
Behalf Of Fatima Chahrour~Vanrise Support
Sent: Wednesday, September 05, 2012 5:19 PM
To: mico...@gmail.com
Cc: 'SIP Router - Kamailio (OpenSER) and SIP
Express Router (SER) - Users Mailing List'
Subject: Re: [SR-Users] FW: Kamailio as
sipRedirect
Hi Daniel,
The call is not
redirected to 1st kamailio, below is the call
trace where 2nd kamailio(192.168.111.139) send
request to 192.168.111.15 which is the gateway in the lcr
rule instead of sending this invite to 192.168.111.138.
U 192.168.111.139:5060 -
192.168.111.15:5060 (shouldn’t
this be 192.168.111.138:5060 instead??)
INVITE sip:22#96135050@192.168.111.15
SIP/2.0.
Record-Route:
sip:192.168.111.139;lr=on.
Record-Route:
sip:192.168.111.138;lr=on.
Via: SIP/2.0/UDP
192.168.111.139;branch=z9hG4bK5757.5b3b1975.0.
Via: SIP/2.0/UDP
192.168.111.138;branch=z9hG4bK5757.50ff9264.0.
Via: SIP/2.0/UDP
192.168.52.139:12650;received=192.168.111.11;branch=z9hG4bK-d8754z-85008205ba3ea607-1---d8754z-;rport=12650.
Max-Forwards: 68.
Contact:
sip:106@192.168.111.11:12650.
To:
"96135050"sip:96135050@192.168.111.138.
From:
"106"sip:106@192.168.111.138;tag=56645478.
Call-ID:
ZjJlYWYzZmJmYmJlMGE0MmYxYWUwYjYwZmY5MmJkMjg..
CSeq: 1 INVITE.
Allow: INVITE, ACK
Re: [SR-Users] FW: Kamailio as sipRedirect
Hello, Not working, I have attached the syslogOnKamailio2-.139.txt generated after a test call from 106 dialing 107. It shows: No Route headers found and '107' Not found in usrloc; what does this means? And is it related? Waiting your feedback. Kind regards, F Chahrour From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Sent: Friday, September 07, 2012 10:33 AM To: Fatima Chahrour~Vanrise Support Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: Re: [SR-Users] FW: Kamailio as sipRedirect Hello, On 9/6/12 6:47 PM, Fatima Chahrour~Vanrise Support wrote: Hi again, I really need your help in my testing lab here!! I have simplified the scenario and changed configs to make it easy pinpoint the redirect issue and then apply on my real scenario: Two Phones(username: 106 and 107) registered on Kamailio1 (192.168.111.138). - Phone 106: dial 107 àOK - Kamailio routing forward call to Kamailio2(192.168.111.139) àOK - Then Kamailio should only receive the call from Kamailio1 and redirect it back to Kamailio1 àNOT OK Note: In my real scenario Kamailio 2 will be responsible for taking the routing decision based on its defined LCR rules and redirect the call with rewritten uri to Kamailio1. Setting in Kamailio2 routing script the following is not working till now: $ru = sip:107@192.168.111.138; send_reply(302, Redirected); the two lines above should be enough for the redirect. I wrote in another email, to preserve the incoming username, use: $ru = sip: + $rU + @192.168.111.138; If it is not working, set debug=3 and watch the log messages (in /var/log/syslog or /var/log/messages) to understand what is not configured properly. Km_append_branch(“uri”); This is not necessary and the parameter has to be a valid SIP URI -- you copied the prototype as it is in docs, defining the format of the function. Cheers, Daniel Kamailio2 is not sending the 302 message and this is my problem. How can I send the 302 message redirect? And where do I have to place or put the related function in the script? Thank you. Kind regards, F.Chahrour U 192.168.111.11:24802 - 192.168.111.138:5060 INVITE sip:107@192.168.111.138 SIP/2.0. Via: SIP/2.0/UDP 192.168.52.139:24802;branch=z9hG4bK-d8754z-7d54de4dd05afd4b-1---d8754z-;rport. Max-Forwards: 70. Contact: sip:106@192.168.111.11:24802. To: 107sip:107@192.168.111.138. From: 106sip:106@192.168.111.138;tag=3d091752. Call-ID: YWUyNTdjYjZhZGZmYmJjMzU0ZGMzNDY1MmFkNWIzNDQ.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: X-Lite release 1100l stamp 47546. Content-Length: 480. . v=0. o=- 4 2 IN IP4 192.168.111.11. s=CounterPath X-Lite 3.0. c=IN IP4 192.168.111.11. t=0 0. m=audio 22038 RTP/AVP 107 119 100 106 0 105 98 8 3 101. a=alt:1 2 : Z3LAcUZR tEUZRy5N 192.168.52.139 22038. a=alt:2 1 : ovb3UWQ7 t62zHtPB 192.168.111.11 22038. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:119 BV32-FEC/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. U 192.168.111.138:5060 - 192.168.111.11:24802 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.52.139:24802;branch=z9hG4bK-d8754z-7d54de4dd05afd4b-1---d8754z-;rport=24802;received=192.168.111.11. To: 107sip:107@192.168.111.138. From: 106sip:106@192.168.111.138;tag=3d091752. Call-ID: YWUyNTdjYjZhZGZmYmJjMzU0ZGMzNDY1MmFkNWIzNDQ.. CSeq: 1 INVITE. Server: kamailio (3.2.4 (i386/linux)). Content-Length: 0. . U 192.168.111.138:5060 - 192.168.111.139:5060 INVITE sip:107@192.168.111.139:5060 SIP/2.0. Record-Route: sip:192.168.111.138;lr=on. Via: SIP/2.0/UDP 192.168.111.138;branch=z9hG4bKd1a7.5a3bc762.0. Via: SIP/2.0/UDP 192.168.52.139:24802;received=192.168.111.11;branch=z9hG4bK-d8754z-7d54de4dd05afd4b-1---d8754z-;rport=24802. Max-Forwards: 69. Contact: sip:106@192.168.111.11:24802. To: 107sip:107@192.168.111.138. From: 106sip:106@192.168.111.138;tag=3d091752. Call-ID: YWUyNTdjYjZhZGZmYmJjMzU0ZGMzNDY1MmFkNWIzNDQ.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: X-Lite release 1100l stamp 47546. Content-Length: 480. . v=0. o=- 4 2 IN IP4 192.168.111.11. s=CounterPath X-Lite 3.0. c=IN IP4 192.168.111.11. t=0 0. m=audio 22038 RTP/AVP 107 119 100 106 0 105 98 8 3 101. a=alt:1 2 : Z3LAcUZR tEUZRy5N 192.168.52.139 22038. a=alt:2 1 : ovb3UWQ7 t62zHtPB 192.168.111.11 22038. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:119 BV32-FEC/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. U 192.168.111.139:5060 - 192.168.111.138:5060 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP
Re: [SR-Users] FW: Kamailio as sipRedirect
Hello, It works now.. It was not an issue in the function you provided but in the presence of route(1) before $ru = sip: + $rU + @192.168.111.138; in the script. And successfully kamailio2 now is able to send the 302 message to .138. Now I’ll work on letting Kamailio1 handle the 302 message in order to make successful call. Thank you Daniel very much for your help. Much appreciated, F Chahrour From: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Fatima Chahrour~Vanrise Support Sent: Friday, September 07, 2012 12:04 PM To: mico...@gmail.com Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: Re: [SR-Users] FW: Kamailio as sipRedirect Hello, Not working, I have attached the syslogOnKamailio2-.139.txt generated after a test call from 106 dialing 107. It shows: No Route headers found and '107' Not found in usrloc; what does this means? And is it related? Waiting your feedback. Kind regards, F Chahrour From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Sent: Friday, September 07, 2012 10:33 AM To: Fatima Chahrour~Vanrise Support Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: Re: [SR-Users] FW: Kamailio as sipRedirect Hello, On 9/6/12 6:47 PM, Fatima Chahrour~Vanrise Support wrote: Hi again, I really need your help in my testing lab here!! I have simplified the scenario and changed configs to make it easy pinpoint the redirect issue and then apply on my real scenario: Two Phones(username: 106 and 107) registered on Kamailio1 (192.168.111.138). - Phone 106: dial 107 àOK - Kamailio routing forward call to Kamailio2(192.168.111.139) àOK - Then Kamailio should only receive the call from Kamailio1 and redirect it back to Kamailio1 àNOT OK Note: In my real scenario Kamailio 2 will be responsible for taking the routing decision based on its defined LCR rules and redirect the call with rewritten uri to Kamailio1. Setting in Kamailio2 routing script the following is not working till now: $ru = sip:107@192.168.111.138; send_reply(302, Redirected); the two lines above should be enough for the redirect. I wrote in another email, to preserve the incoming username, use: $ru = sip: + $rU + @192.168.111.138; If it is not working, set debug=3 and watch the log messages (in /var/log/syslog or /var/log/messages) to understand what is not configured properly. Km_append_branch(“uri”); This is not necessary and the parameter has to be a valid SIP URI -- you copied the prototype as it is in docs, defining the format of the function. Cheers, Daniel Kamailio2 is not sending the 302 message and this is my problem. How can I send the 302 message redirect? And where do I have to place or put the related function in the script? Thank you. Kind regards, F.Chahrour ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] FW: Kamailio as sipRedirect
Hi again,
I really need your help in my testing lab here!!
I have simplified the scenario and changed configs to make it easy pinpoint the
redirect issue and then apply on my real scenario:
Two Phones(username: 106 and 107) registered on Kamailio1 (192.168.111.138).
- Phone 106: dial 107 àOK
- Kamailio routing forward call to Kamailio2(192.168.111.139) àOK
- Then Kamailio should only receive the call from Kamailio1 and
redirect it back to Kamailio1 àNOT OK
Note: In my real scenario Kamailio 2 will be responsible for taking the routing
decision based on its defined LCR rules and redirect the call with rewritten
uri to Kamailio1.
Setting in Kamailio2 routing script the following is not working till now:
$ru = sip:107@192.168.111.138;
send_reply(302, Redirected);
Km_append_branch(“uri”);
Kamailio2 is not sending the 302 message and this is my problem.
How can I send the 302 message redirect? And where do I have to place or put
the related function in the script?
Thank you.
Kind regards,
F.Chahrour
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Fatima
Chahrour~Vanrise Support
Sent: Thursday, September 06, 2012 4:40 PM
To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List'; mico...@gmail.com
Subject: Re: [SR-Users] FW: Kamailio as sipRedirect
Dear Daniel,
Attached is my config file.
Is it correct to set $ru=”sip:.*@192.168.111.138; in case I need any number to
be redirected to 192.168.111.138.
Thanks.
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Fatima
Chahrour~Vanrise Support
Sent: Wednesday, September 05, 2012 5:19 PM
To: mico...@gmail.com
Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List'
Subject: Re: [SR-Users] FW: Kamailio as sipRedirect
Hi Daniel,
The call is not redirected to 1st kamailio, below is the call trace where 2nd
kamailio(192.168.111.139) send request to 192.168.111.15 which is the gateway
in the lcr rule instead of sending this invite to 192.168.111.138.
U 192.168.111.139:5060 - 192.168.111.15:5060 (shouldn’t this be
192.168.111.138:5060 instead??)
INVITE sip:22#96135050@192.168.111.15 SIP/2.0.
Record-Route: sip:192.168.111.139;lr=on.
Record-Route: sip:192.168.111.138;lr=on.
Via: SIP/2.0/UDP 192.168.111.139;branch=z9hG4bK5757.5b3b1975.0.
Via: SIP/2.0/UDP 192.168.111.138;branch=z9hG4bK5757.50ff9264.0.
Via: SIP/2.0/UDP
192.168.52.139:12650;received=192.168.111.11;branch=z9hG4bK-d8754z-85008205ba3ea607-1---d8754z-;rport=12650.
Max-Forwards: 68.
Contact: sip:106@192.168.111.11:12650.
To: 96135050sip:96135050@192.168.111.138.
From: 106sip:106@192.168.111.138;tag=56645478.
Call-ID: ZjJlYWYzZmJmYmJlMGE0MmYxYWUwYjYwZmY5MmJkMjg..
CSeq: 1 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO.
Content-Type: application/sdp.
User-Agent: X-Lite release 1100l stamp 47546.
Content-Length: 477.
This is my Kamailio routing logic configured on the 1st Kamailio to forward any
call from 1st kamialio to 2nd kamailio:
if (is_method(INVITE))
{
Sethostport(“192.168.111.139:5060”);
Append_branch();
}
My problem is hereà This is my Kamailio routing configured on 2nd kamailio
which I want from it to send back the call, with lcr_gateway already defined,
to 1st Kamailio:
if (is_method(INVITE))
{
if (!load_gws(1)) {
sl_send_reply(503, Unable to load gateways);
exit;
}
if(!next_gw()){
sl_send_reply(503, Unable to find a gateway);
exit;}
route(1);
$ru=sip:.*@192.168.111.138;
send_reply(302,rEdIrEcTeD);
km_append_branch(uri);
}
Am looking through many documents but can’t achieve my target, I want the 2nd
Kamailio, which has LCR configured on it, to send back to 1st Kamailio the
gateway from lcr_gateways with 302 redirect message.
My questions are:
How to achieve this?
Do you suggest any document that help?
Do I need to load any specific module?
Thanks,
F Chahrour
From: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Sent: Wednesday, September 05, 2012 9:38 AM
To: Fatima Chahrour~Vanrise Support
Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List'
Subject: Re: [SR-Users] FW: Kamailio as sipRedirect
Hello,
You set r-uri ($ru) to an IP address only -- it has to be a full SIP URI.
Cheers,
Daniel
On 9/4/12 5:34 PM, Fatima Chahrour~Vanrise Support wrote:
Hello,
Thanks for your reply.. am able to make the call reach from 3CX to Kamailio
using the src_ip, thank you, but still not able
Re: [SR-Users] FW: Kamailio as sipRedirect
Hello,
You set r-uri ($ru) to an IP address only -- it has to be a full SIP
URI.
Cheers,
Daniel
On 9/4/12 5:34 PM, Fatima
Chahrour~Vanrise Support wrote:
Hello,
Thanks for your
reply.. am able to make the call reach from 3CX to Kamailio
using the src_ip, thank you, but still not able to redirect
the call invite from Kamailio to 3CX and send the 302
message.
Kamailio
192.168.111.139
Phone A (192.168.111.11)
302
Phone B
(192.168.111.10)
Herby what am
putting in my routing:
If (is_method("INVITE"))
{ if(!load_gateway("1")) {
sl_send_reply("503", "unable to load
gateways");
Exit; }
{ if(! next_gateway()) {
sl_send_reply("503", "unable to find
gateways");
Exit; }
Route(1);
$ru = "192.168.111.138";
send_reply("302", "Redirected";
Km_append_branch(“uri”);
Exit;
What am I doing
is wrong in the above script? And causing “call failed”!
attached ‘test.txt’ the ngrep capture if needed.
Your help is
highly appreciated.
Kind Regards,
F Chahrour
From:
sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On
Behalf Of Daniel-Constantin Mierla
Sent: Thursday, August 30, 2012 10:56 AM
To: Fatima Chahrour
Cc: SIP Router - Kamailio (OpenSER) and SIP
Express Router (SER) - Users Mailing List
Subject: Re: [SR-Users] FW: Kamailio as
sipRedirect
Hello,
if you know the IP address of 3CX, then you can skip
authentication for that, like:
if(src_ip==__3cx_ip__) {
# don't do authentication for the traffic coming here
}
You can use permission module to store the a list of ip
addresses and then use functions from that module to match
against source ip of requests, as alternative to above,
providing more flexibility when dealing with lot of trusted ip
addresses.
Cheers,
Daniel
On 8/29/12 9:15 PM, Fatima Chahrour
wrote:
Hello Danial,
Hereby details:
My current working 1st scenario is a
call generated from A-party x-lite registered on
kamalio, this call is routed based on kamailio lcr
routing module which add prefix and send prefix to 3CX
then 3CX forward the call to B-party x-lite, registered
on 3CX, successfully.
In kamailio.cfg routing logic I only
used :
...
If (is_method("INVITE"))
Re: [SR-Users] FW: Kamailio as sipRedirect
Hi Daniel,
The call is not redirected to 1st kamailio, below is the call trace where 2nd
kamailio(192.168.111.139) send request to 192.168.111.15 which is the gateway
in the lcr rule instead of sending this invite to 192.168.111.138.
U 192.168.111.139:5060 - 192.168.111.15:5060 (shouldn’t this be
192.168.111.138:5060 instead??)
INVITE sip:22#96135050@192.168.111.15 SIP/2.0.
Record-Route: sip:192.168.111.139;lr=on.
Record-Route: sip:192.168.111.138;lr=on.
Via: SIP/2.0/UDP 192.168.111.139;branch=z9hG4bK5757.5b3b1975.0.
Via: SIP/2.0/UDP 192.168.111.138;branch=z9hG4bK5757.50ff9264.0.
Via: SIP/2.0/UDP
192.168.52.139:12650;received=192.168.111.11;branch=z9hG4bK-d8754z-85008205ba3ea607-1---d8754z-;rport=12650.
Max-Forwards: 68.
Contact: sip:106@192.168.111.11:12650.
To: 96135050sip:96135050@192.168.111.138.
From: 106sip:106@192.168.111.138;tag=56645478.
Call-ID: ZjJlYWYzZmJmYmJlMGE0MmYxYWUwYjYwZmY5MmJkMjg..
CSeq: 1 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO.
Content-Type: application/sdp.
User-Agent: X-Lite release 1100l stamp 47546.
Content-Length: 477.
This is my Kamailio routing logic configured on the 1st Kamailio to forward any
call from 1st kamialio to 2nd kamailio:
if (is_method(INVITE))
{
Sethostport(“192.168.111.139:5060”);
Append_branch();
}
My problem is hereà This is my Kamailio routing configured on 2nd kamailio
which I want from it to send back the call, with lcr_gateway already defined,
to 1st Kamailio:
if (is_method(INVITE))
{
if (!load_gws(1)) {
sl_send_reply(503, Unable to load gateways);
exit;
}
if(!next_gw()){
sl_send_reply(503, Unable to find a gateway);
exit;}
route(1);
$ru=sip:.*@192.168.111.138;
send_reply(302,rEdIrEcTeD);
km_append_branch(uri);
}
Am looking through many documents but can’t achieve my target, I want the 2nd
Kamailio, which has LCR configured on it, to send back to 1st Kamailio the
gateway from lcr_gateways with 302 redirect message.
My questions are:
How to achieve this?
Do you suggest any document that help?
Do I need to load any specific module?
Thanks,
F Chahrour
From: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Sent: Wednesday, September 05, 2012 9:38 AM
To: Fatima Chahrour~Vanrise Support
Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List'
Subject: Re: [SR-Users] FW: Kamailio as sipRedirect
Hello,
You set r-uri ($ru) to an IP address only -- it has to be a full SIP URI.
Cheers,
Daniel
On 9/4/12 5:34 PM, Fatima Chahrour~Vanrise Support wrote:
Hello,
Thanks for your reply.. am able to make the call reach from 3CX to Kamailio
using the src_ip, thank you, but still not able to redirect the call invite
from Kamailio to 3CX and send the 302 message.
Kamailio instead of 3CX
192.168.111.138
Kamailio
192.168.111.139
Phone A (192.168.111.11)
302
Phone B (192.168.111.10)
Herby what am putting in my routing:
If (is_method(INVITE))
{ if(!load_gateway(1)) {
sl_send_reply(503, unable to load gateways);
Exit; }
{ if(! next_gateway()) {
sl_send_reply(503, unable to find gateways);
Exit; }
Route(1);
$ru = 192.168.111.138;
send_reply(302, Redirected;
Km_append_branch(“uri”);
Exit;
What am I doing is wrong in the above script? And causing “call failed”!
attached ‘test.txt’ the ngrep capture if needed.
Your help is highly appreciated.
Kind Regards,
F Chahrour
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Daniel-Constantin
Mierla
Sent: Thursday, August 30, 2012 10:56 AM
To: Fatima Chahrour
Cc: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List
Subject: Re: [SR-Users] FW: Kamailio as sipRedirect
Hello,
if you know the IP address of 3CX, then you can skip authentication for that,
like:
if(src_ip==__3cx_ip__) {
# don't do authentication for the traffic coming here
}
You can use permission module to store the a list of ip addresses and then use
functions from that module to match against source ip of requests, as
alternative to above, providing more flexibility when dealing with lot of
trusted ip addresses.
Cheers,
Daniel
On 8/29/12 9:15 PM, Fatima Chahrour wrote:
Hello Danial,
Hereby details:
My current working 1st scenario is a call generated from A-party x-lite
registered on kamalio, this call is routed based on kamailio lcr routing module
which add prefix and send prefix to 3CX then 3CX forward the call to B-party
x-lite, registered on 3CX, successfully.
In kamailio.cfg routing logic I only used :
...
If (is_method
Re: [SR-Users] FW: Kamailio as sipRedirect
Hello,
Thanks for your reply.. am able to make the call reach from 3CX to Kamailio
using the src_ip, thank you, but still not able to redirect the call invite
from Kamailio to 3CX and send the 302 message.
Kamailio
192.168.111.139
Phone A (192.168.111.11)
302
Phone B (192.168.111.10)
Herby what am putting in my routing:
If (is_method(INVITE))
{ if(!load_gateway(1)) {
sl_send_reply(503, unable to load gateways);
Exit; }
{ if(! next_gateway()) {
sl_send_reply(503, unable to find gateways);
Exit; }
Route(1);
$ru = 192.168.111.138;
send_reply(302, Redirected;
Km_append_branch(“uri”);
Exit;
What am I doing is wrong in the above script? And causing “call failed”!
attached ‘test.txt’ the ngrep capture if needed.
Your help is highly appreciated.
Kind Regards,
F Chahrour
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Daniel-Constantin
Mierla
Sent: Thursday, August 30, 2012 10:56 AM
To: Fatima Chahrour
Cc: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List
Subject: Re: [SR-Users] FW: Kamailio as sipRedirect
Hello,
if you know the IP address of 3CX, then you can skip authentication for that,
like:
if(src_ip==__3cx_ip__) {
# don't do authentication for the traffic coming here
}
You can use permission module to store the a list of ip addresses and then use
functions from that module to match against source ip of requests, as
alternative to above, providing more flexibility when dealing with lot of
trusted ip addresses.
Cheers,
Daniel
On 8/29/12 9:15 PM, Fatima Chahrour wrote:
Hello Danial,
Hereby details:
My current working 1st scenario is a call generated from A-party x-lite
registered on kamalio, this call is routed based on kamailio lcr routing module
which add prefix and send prefix to 3CX then 3CX forward the call to B-party
x-lite, registered on 3CX, successfully.
In kamailio.cfg routing logic I only used :
...
If (is_method(INVITE))
{ if(!load_gateway(1)) {
sl_send_reply(503, unable to load gateways);
Exit; }
{ if(! next_gateway()) {
sl_send_reply(503, unable to find gateways);
Exit; }
Route(1);
Now, my target is to register also the A-party x-lite to 3CX where I added a
rule in 3CX to forward the call comming to kamailio to take the route desicion
and act.
So, here I need to know how can I tell kamailio to accept the call from 3CX?
this is what I meant by configuring 3CX on kamailio.
I have weakness in the routing logic configuration, could you please help in
this too! What is the required replacement on the above to be done?
Moreover, I appreciate if you provide with docs or tutorials that can help me
understand more kamailio's routing logic functions!
Much appreciated.
Thanks in advance.
F Chahrour
Sent from iPhone
On Aug 29, 2012, at 9:21 PM, Daniel-Constantin Mierla mico...@gmail.com wrote:
Hello,
On 8/29/12 3:31 PM, Fatima Chahrour~Vanrise Support wrote:
Dears,
I have an installed 3CX where I want to generate a call from it to the Kamailo
to get a list of suppliers to terminate the call to.
Thus, the Kamailio should act as a sip redirect to the 3CX (for routing
decision).
Have you tried such configuration before? And how do I configure the 3CX on the
Kamailio?
I don't understand what you expect from 'configure the 3CX on the Kamailio',
can you add more details ?
For sending a redirect, practically you have to update request uri and then
send the 3xx reply, like:
$ru = sip:newnum...@newdomain.com sip:newnum...@newdomain.com;
send_reply(302, Redirected;
If you need to put more contacts in the redirect reply, then add:
km_append_branch(uri);
uri parameter can contain variables.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda http://twitter.com/#%21/miconda -
http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
image001.pngimage005.pngimage006.pnginterface: eth0 (192.168.111.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )
U 192.168.111.138:5060 - 192.168.111.139:5060
INVITE sip:961325252525@192.168.111.139:5060 SIP/2.0.
Record-Route: sip:192.168.111.138;lr=on.
Via: SIP/2.0/UDP 192.168.111.138;branch=z9hG4bK9c47.a9dbeca3.0.
Via: SIP/2.0/UDP
192.168.52.139:12382;received=192.168.111.11;branch=z9hG4bK-d8754z-6b29401c46626f6f-1---d8754z-;rport=12382.
Max-Forwards: 69.
Contact: sip:106@192.168.111.11:12382.
To: 961325252525sip:961325252525@192.168.111.138.
From: 106sip:106@192.168.111.138;tag=ac5dc85f.
Call-ID
Re: [SR-Users] FW: Kamailio as sipRedirect
Hello,
if you know the IP address of 3CX, then you can skip authentication for
that, like:
if(src_ip==__3cx_ip__) {
# don't do authentication for the traffic coming here
}
You can use permission module to store the a list of ip addresses and
then use functions from that module to match against source ip of
requests, as alternative to above, providing more flexibility when
dealing with lot of trusted ip addresses.
Cheers,
Daniel
On 8/29/12 9:15 PM, Fatima Chahrour wrote:
Hello Danial,
Hereby details:
My current working 1st scenario is a call generated from A-party
x-lite registered on kamalio, this call is routed based on kamailio
lcr routing module which add prefix and send prefix to 3CX then 3CX
forward the call to B-party x-lite, registered on 3CX, successfully.
In kamailio.cfg routing logic I only used :
...
If (is_method(INVITE))
{ if(!load_gateway(1)) {
sl_send_reply(503, unable to load gateways);
Exit; }
{ if(! next_gateway()) {
sl_send_reply(503, unable to find gateways);
Exit; }
Route(1);
Now, my target is to register also the A-party x-lite to 3CX where I
added a rule in 3CX to forward the call comming to kamailio to take
the route desicion and act.
So, here I need to know how can I tell kamailio to accept the call
from 3CX? this is what I meant by configuring 3CX on kamailio.
I have weakness in the routing logic configuration, could you please
help in this too! What is the required replacement on the above to be
done?
Moreover, I appreciate if you provide with docs or tutorials that can
help me understand more kamailio's routing logic functions!
Much appreciated.
Thanks in advance.
F Chahrour
Sent from iPhone
On Aug 29, 2012, at 9:21 PM, Daniel-Constantin Mierla
mico...@gmail.com mailto:mico...@gmail.com wrote:
Hello,
On 8/29/12 3:31 PM, Fatima Chahrour~Vanrise Support wrote:
Dears,
I have an installed 3CX where I want to generate a call from it to
the Kamailo to get a list of suppliers to terminate the call to.
Thus, the Kamailio should act as a sip redirect to the 3CX (for
routing decision).
Have you tried such configuration before? And how do I configure the
3CX on the Kamailio?
I don't understand what you expect from 'configure the 3CX on the
Kamailio', can you add more details ?
For sending a redirect, practically you have to update request uri
and then send the 3xx reply, like:
$ru = sip:newnum...@newdomain.com;
send_reply(302, Redirected;
If you need to put more contacts in the redirect reply, then add:
km_append_branch(uri);
uri parameter can contain variables.
Cheers,
Daniel
--
Daniel-Constantin Mierla -http://www.asipto.com
http://twitter.com/#!/miconda http://twitter.com/#%21/miconda
-http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] FW: Kamailio as sipRedirect
Hello, On 8/29/12 3:31 PM, Fatima Chahrour~Vanrise Support wrote: Dears, I have an installed 3CX where I want to generate a call from it to the Kamailo to get a list of suppliers to terminate the call to. Thus, the Kamailio should act as a sip redirect to the 3CX (for routing decision). Have you tried such configuration before? And how do I configure the 3CX on the Kamailio? I don't understand what you expect from 'configure the 3CX on the Kamailio', can you add more details ? For sending a redirect, practically you have to update request uri and then send the 3xx reply, like: $ru = sip:newnum...@newdomain.com; send_reply(302, Redirected; If you need to put more contacts in the redirect reply, then add: km_append_branch(uri); uri parameter can contain variables. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] FW: Kamailio as sipRedirect
Hello Danial,
Hereby details:
My current working 1st scenario is a call generated from A-party x-lite
registered on kamalio, this call is routed based on kamailio lcr routing module
which add prefix and send prefix to 3CX then 3CX forward the call to B-party
x-lite, registered on 3CX, successfully.
In kamailio.cfg routing logic I only used :
...
If (is_method(INVITE))
{ if(!load_gateway(1)) {
sl_send_reply(503, unable to load gateways);
Exit; }
{ if(! next_gateway()) {
sl_send_reply(503, unable to find gateways);
Exit; }
Route(1);
Now, my target is to register also the A-party x-lite to 3CX where I added a
rule in 3CX to forward the call comming to kamailio to take the route desicion
and act.
So, here I need to know how can I tell kamailio to accept the call from 3CX?
this is what I meant by configuring 3CX on kamailio.
I have weakness in the routing logic configuration, could you please help in
this too! What is the required replacement on the above to be done?
Moreover, I appreciate if you provide with docs or tutorials that can help me
understand more kamailio's routing logic functions!
Much appreciated.
Thanks in advance.
F Chahrour
Sent from iPhone
On Aug 29, 2012, at 9:21 PM, Daniel-Constantin Mierla mico...@gmail.com wrote:
Hello,
On 8/29/12 3:31 PM, Fatima Chahrour~Vanrise Support wrote:
Dears,
I have an installed 3CX where I want to generate a call from it to the
Kamailo to get a list of suppliers to terminate the call to.
Thus, the Kamailio should act as a sip redirect to the 3CX (for routing
decision).
Have you tried such configuration before? And how do I configure the 3CX on
the Kamailio?
I don't understand what you expect from 'configure the 3CX on the Kamailio',
can you add more details ?
For sending a redirect, practically you have to update request uri and then
send the 3xx reply, like:
$ru = sip:newnum...@newdomain.com;
send_reply(302, Redirected;
If you need to put more contacts in the redirect reply, then add:
km_append_branch(uri);
uri parameter can contain variables.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
