Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route - FreeSWITCH-users

31 Aug 2009


      Now it is correct.

Thank you for your time.
...
-------- Оригинално писмо --------
От:  rod 
Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Събота, 2009, Август 29 08:43:24 EEST
...
Ok,
I found what's happening. I probably did some change on the wiki to 
start reflect the new configuration, without having time enough to check 
the configuration. There is a mistake in your dialplan configuration. 
You should put this instead:
wrong line:
please correct with:
This should match PEER_01 in dialplan instead of trying matching 
France@PEER_01.
Let me know if this is right now.
rod
Hristo Benev a écrit :
...
Hello Rod,
I did the change.
Here is extract of console:
-----------------------------------------------------------------------------------------
variable_continue_on_fail: [true]
variable_sip_h_X-ROUTE: [LOOKUP]
variable_export_vars: [sip_h_X-ROUTE]
variable_signal_bond: [a66d089a-93ee-11de-ab7a-094dff0ede20]
variable_sip_redirect_contact_0: [sip:France@PEER_01]
variable_sip_redirected_to: [sip:France@PEER_01]
variable_sip_redirect_contact_user_0: [France]  variable_sip_redirect_contact_host_0: [PEER_01]
variable_sip_redirect_dialstring_0: [sofia/internal/sip:France@PEER_01]
variable_sip_redirect_dialstring: [sofia/internal/sip:France@PEER_01]
variable_proto_specific_hangup_cause: [sip:503]
variable_sip_hangup_phrase: [DNS Error]
variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE]
variable_ROUTE_GW: [France]
variable_AREA: [France]
variable_current_application: [info]
------------------------------------------------------------------------------------------
I have different value it is actually the description field as shown here:
--------------------------------
 /opt/kamailio/sbin/kamctl cr show
cr carrier names
+----+---------+
| id | carrier |
+----+---------+
|  1 | default |
+----+---------+
cr domain names
+----+---------+
| id | domain  |
+----+---------+
|  1 | default |
+----+---------+
cr routes
+----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
| id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description |
+----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
|  1 |       1 |      1 | 1000        |     0 |    0 |    1 |     0 | PEER_01      |                |                | France      |
+----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
-----------------------------------------
And here is what I have in kamailio:
-------------------------------------------------------------------------------------
####### Routing Logic ########
# main request routing logic
route{
if (!mf_process_maxfwd_header("10")) {
                sl_send_reply("483","Too Many Hops");
                exit;
        }
t_check_trans();
if ($rU==NULL) {
                # request with no Username in RURI
                sl_send_reply("484","Address Incomplete");
                exit;
        }
# LOOKUP ROUTE TABLE WHEN ASKED BY HEADER: X-ROUTE:LOOKUP
        if (is_method("INVITE") && $hdr(X-ROUTE)=="LOOKUP"){
                if(!cr_route("default", "default", "$rU", "$rU", "call_id","$avp(s:route_desc)")){
                     #xlog("ROUTING FAILED: no route found for $rU");
                     sl_send_reply("604", "Unable to route this call");
                     exit;
                } else {
                     xlog("LOOKUP FOUND: $rd $avp(s:route_desc)");
                     avp_pushto("$ru/username", "$avp(s:route_desc)");
                     sl_send_reply("302", "$rd");
                     exit;
                }
        }
}
----------------------------------------------------------------------------------
Another question...
In that part of FreeSwitch dialplan.xml
-----------------------------------
-----------------------------------
X-ROUTE maybe should be replaced with ROUTE_GW otherwise there is no way the regex is true.
As for thanks - for sure by default they are also for the developers of both apps.
I'm new in freeSwitch and/or Kamailio for now I'm still testing and learning so it will be nice to have something working to start with.
...
-------- Оригинално писмо --------
От:  rod 
Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Петък, 2009, Август 28 09:54:00 EEST
...
Hello,
the trace seems good.
If you check the answer from Kamailio, you'll see that Kamailio answers 
with "302 PEER_01".
As Michael Collins stated before, you can get the variable containing 
"PEER_01", then this variable is stored in a custom variable.
In your dialplan, may you please add:
, just before the transfer line, eg:
Using application Info, you'll see on the console (or CDR) the list of 
variables used for this call.  You should see the content of 
"${sip_redirect_contact_user_0}" that should contain  the value 
"PEER_01". Please check this and let me know.
For the new configuration file, no problem for sharing. But as I wrote 
on the wiki page, I worked on this setup cause LCR module was not 
available when I start working on FS, nor I'm a good programmer to write 
a server side HTTP script (used by xml_curl) that could scale to my needs.
I enhanced a bit this configuration with support for fallback routing, 
almost realtime graph (every minutes using www.cacti.net and some 
functions in FS like limit_hash) of number of concurrent calls per AREA, 
PEER...
Thanks for "the good tutorial", but don't forget the dev team who did 
this great product ;-)
rod.
Hristo Benev a écrit :
...
I assume you asked for port 5062 since I do not have any traffic on 5060 (I have one IP and my internal sip port is 5090 and external 5080).
If you need additional info I'll provide it.
Here is trace:
ngrep -d any -nn -i '1000' port 5062 -W byline
interface: any
filter: (ip or ip6) and ( port 5062 )
match: 1000
#
U 10.10.10.10:5090 -> 127.0.0.1:5062
INVITE sip:1000@127.0.0.1:5062 SIP/2.0.
Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
Max-Forwards: 69.
From: "Extension 1001" ;tag=978g69jZaFpBD.
To: .
Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
CSeq: 119574771 INVITE.
Contact: .
User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 429.
X-ROUTE: LOOKUP.
Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1251355692 1251355693 IN IP4 10.10.10.10.
s=FreeSWITCH.
c=IN IP4 10.10.10.10.
t=0 0.
m=audio 30522 RTP/AVP 0 115 107 9 8 3 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:115 G7221/32000.
a=fmtp:115 bitrate=48000.
a=rtpmap:107 G7221/16000.
a=fmtp:107 bitrate=32000.
a=rtpmap:9 G722/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.
#
U 127.0.0.1:5062 -> 10.10.10.10:5090
SIP/2.0 302 PEER_01.
Via: SIP/2.0/UDP 10.10.10.10:5090;rport=5090;branch=z9hG4bKtrDyU1US5X5tj.
From: "Extension 1001" ;tag=978g69jZaFpBD.
To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
CSeq: 119574771 INVITE.
Contact: sip:France@PEER_01.
Server: Kamailio (1.5.2-notls (i386/linux)).
Content-Length: 0.
.
#
U 10.10.10.10:5090 -> 127.0.0.1:5062
ACK sip:1000@127.0.0.1:5062 SIP/2.0.
Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
Max-Forwards: 69.
From: "Extension 1001" ;tag=978g69jZaFpBD.
To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
CSeq: 119574771 ACK.
Content-Length: 0.
.
...
-------- Оригинално писмо --------
От:  Hristo Benev 
Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Четвъртък, 2009, Август 27 14:58:53 EEST
...
Bojnour,
I'll send a trace ASAP.
What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config)
I'm really interested to see the updates of the project.
Thank you for the good tutorial.
Hristo
...
-------- Оригинално писмо --------
От:  rod 
Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Четвъртък, 2009, Август 27 09:41:05 EEST
...
Hi Hristo,
I'm the author of this setup and wiki page. I did a lot of modifications 
on this setup (alternative routing if failure essentially) but don't 
have too much time to update the wiki.
May you please send me an ngrep trace when you call 1000:
ngrep -d any -nn -i '1000' port 5060 -W byline
I will check what's happening.
Do you have an entry for 1000 in your mysql database ?
regards,
rod
Hristo Benev a écrit :
>  
> It seems that the problem is on kamailio configuration.
> Will ask on their list.
>
> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.
>
> I have it setup as friend in asterisk, but still ???
>
> Any ideas?
>
> Thanks,
>
>
>  >-------- Оригинално писмо --------
>  >От:  Hristo Benev 
>  >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >До: freeswitch-users@lists.freeswitch.org
>  >Изпратено на: Сряда, 2009, Август 26 22:02:13 EEST
>
>  > I think that the problem is here:
>  >-------------------------
>  >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
>  >Dialplan: sofia/internal/1001@209.71.254.33 parsing [ROUTING->PEER_01] continue=false
>  >Dialplan: sofia/internal/1001@209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
>  >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
>  >--------------------------
>  >
>  >Actually Regex FAIL
>  >
>  >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
>  >Here is my  default.xml:
>  >----------------
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>  >      
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>  >--------------------------
>  >
>  >
>  >
>  > >-------- Оригинално писмо --------
>  > >От:  Brian West 
>  > >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  > >До: freeswitch-users@lists.freeswitch.org
>  > >Изпратено на: Сряда, 2009, Август 26 19:47:37 EEST
>  >
>  > >We do not blindly follow 302's as that is a dangerous thing to do. 
>  >You have to process all 302's in the dialplan. 
>  >Set this on your sofia profile
>  >You can set these variables sip_redirect_profile,
>  >sip_redirect_context,
>  >sip_redirect_dialplan,
>  >When a redirect happens you get these variables - sip_redirect_contact_%d,
>  >sip_redirected_to,
>  >sip_redirect_contact_user_%d,
>  >sip_redirect_contact_host_%d,
>  >sip_redirect_contact_params_%d,
>  >sip_redirect_dialstring_%d,
>  >sip_redirect_dialstring,
>  >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
>  >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
>  > >
>  >
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Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route

Hristo Benev

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