Re: [Freeswitch-users] Scanning my firewall for open UDP ports?
Just for your information there is a version of nmap for windows. So you can do the test from your desktop. Оригинално писмо От: Fred-145 Относно: Re: [Freeswitch-users] Scanning my firewall for open UDP ports? До: freeswitch-users@lists.freeswitch.org Изпратено на: Четвъртък, 2009, Декември 17 14:54:49 EET I don't have access to a remote computer from which I could log on and run nmap. I'll see if I can get a shell access somewhere. Thank you. -- View this message in context: http://old.nabble.com/Scanning-my-firewall-for-open-UDP-ports--tp26808383p26827581.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] mod_cdr_csv and mysql
Hello, I saw an sql option in mod_cdr_csv. For my surprise it wrote sql code in Master.csv file instead of recording to mysql database (already setup as ODBC) Is that normal or I'm missing something? I read on the wiki that there is additional perl script to load csv to mysql. Should I do it in that way? Thanks, ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
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
fra...@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:fra...@peer_01]
variable_sip_redirected_to: [sip:fra...@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:fra...@peer_01]
variable_sip_redirect_dialstring: [sofia/internal/sip:fra...@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
Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
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:1...@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:fra...@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:1...@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
Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
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/1...@209.71.254.33 parsing [ROUTING-PEER_01]
continue=false
Dialplan: sofia/internal/1...@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:
--
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
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:1...@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 sip:1...@10.10.10.10;tag=978g69jZaFpBD.
To: sip:1...@127.0.0.1:5062.
Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
CSeq: 119574771 INVITE.
Contact: sip:mod_so...@10.10.10.10:5090.
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
sip:1...@10.10.10.10;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 sip:1...@10.10.10.10;tag=978g69jZaFpBD.
To: sip:1...@127.0.0.1:5062;tag=458fb4012080e656b6742c09466dabcd.31c8.
Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
CSeq: 119574771 INVITE.
Contact: sip:fra...@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:1...@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 sip:1...@10.10.10.10;tag=978g69jZaFpBD.
To: sip:1...@127.0.0.1:5062;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/1...@209.71.254.33 parsing [ROUTING-PEER_01]
continue=false
Dialplan: sofia/internal/1...@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 ${sip_h_X-ROUTE} should be
PEER_01 for success?
Here is my default.xml
[Freeswitch-users] freeswitch as SBC and kamailio - no route
Hello
I followed the tutorial
http://wiki.freeswitch.org/wiki/SBC_Setup
I have following problem when I dial 1000 Kamalio reports 302, but freeswitch
does not route
Where to look for problems?
Here is the debug:
2009-08-26 20:21:42.332154 [INFO] mod_dialplan_xml.c:315 Processing 1001-1000
in context default
Dialplan: sofia/internal/1...@10.10.10.10 parsing [default-LOOKUP_ROUTE]
continue=false
Dialplan: sofia/internal/1...@10.10.10.10 Regex (PASS) [LOOKUP_ROUTE]
destination_number(1000) =~ /(\d+)$/ break=on-false
Dialplan: sofia/internal/1...@10.10.10.10 Action set(hangup_after_bridge=true)
Dialplan: sofia/internal/1...@10.10.10.10 Action set(continue_on_fail=true)
Dialplan: sofia/internal/1...@10.10.10.10 Action export(sip_h_X-ROUTE=LOOKUP)
Dialplan: sofia/internal/1...@10.10.10.10 Action
bridge(sofia/internal/${destination_numb...@127.0.0.1:5062)
Dialplan: sofia/internal/1...@10.10.10.10 Action
set(ROUTE_GW=${sip_redirect_contact_user_0})
Dialplan: sofia/internal/1...@10.10.10.10 Action
set(AREA=${sip_redirect_contact_user_0})
Dialplan: sofia/internal/1...@10.10.10.10 Action transfer(${destination_number}
XML ROUTING)
2009-08-26 20:21:42.334579 [DEBUG] switch_core_state_machine.c:114
(sofia/internal/1...@10.10.10.10) State Change CS_ROUTING - CS_EXECUTE
2009-08-26 20:21:42.334579 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1...@10.10.10.10 [BREAK]
2009-08-26 20:21:42.334579 [DEBUG] switch_core_state_machine.c:484
(sofia/internal/1...@10.10.10.10) State ROUTING going to sleep
2009-08-26 20:21:42.335292 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1...@10.10.10.10) Running State Change CS_EXECUTE
2009-08-26 20:21:42.335292 [DEBUG] switch_core_state_machine.c:491
(sofia/internal/1...@10.10.10.10) State EXECUTE
2009-08-26 20:21:42.335292 [DEBUG] mod_sofia.c:173
sofia/internal/1...@10.10.10.10 SOFIA EXECUTE
2009-08-26 20:21:42.335292 [DEBUG] switch_core_state_machine.c:151
sofia/internal/1...@10.10.10.10 Standard EXECUTE
EXECUTE sofia/internal/1...@10.10.10.10 set(hangup_after_bridge=true)
2009-08-26 20:21:42.336509 [DEBUG] mod_dptools.c:748
sofia/internal/1...@10.10.10.10 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1...@10.10.10.10 set(continue_on_fail=true)
2009-08-26 20:21:42.337353 [DEBUG] mod_dptools.c:748
sofia/internal/1...@10.10.10.10 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/1...@10.10.10.10 export(sip_h_X-ROUTE=LOOKUP)
2009-08-26 20:21:42.338352 [DEBUG] mod_dptools.c:886 EXPORT
[sip_h_X-ROUTE]=[LOOKUP]
EXECUTE sofia/internal/1...@10.10.10.10
bridge(sofia/internal/1...@127.0.0.1:5062)
2009-08-26 20:21:42.339309 [NOTICE] switch_channel.c:602 New Channel
sofia/internal/1...@127.0.0.1:5062 [8a588e6a-925c-11de-85dd-15dc0a06983f]
2009-08-26 20:21:42.339309 [DEBUG] mod_sofia.c:2811
(sofia/internal/1...@127.0.0.1:5062) State Change CS_NEW - CS_INIT
2009-08-26 20:21:42.340376 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1...@127.0.0.1:5062 [BREAK]
2009-08-26 20:21:42.341175 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1...@127.0.0.1:5062) Running State Change CS_INIT
2009-08-26 20:21:42.341175 [DEBUG] switch_core_state_machine.c:481
(sofia/internal/1...@127.0.0.1:5062) State INIT
2009-08-26 20:21:42.341175 [DEBUG] mod_sofia.c:83
sofia/internal/1...@127.0.0.1:5062 SOFIA INIT
2009-08-26 20:21:42.344378 [DEBUG] mod_sofia.c:111
(sofia/internal/1...@127.0.0.1:5062) State Change CS_INIT - CS_ROUTING
2009-08-26 20:21:42.344378 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1...@127.0.0.1:5062 [BREAK]
2009-08-26 20:21:42.344378 [DEBUG] switch_core_state_machine.c:481
(sofia/internal/1...@127.0.0.1:5062) State INIT going to sleep
2009-08-26 20:21:42.344378 [DEBUG] sofia.c:3289 Channel
sofia/internal/1...@127.0.0.1:5062 entering state [calling][0]
2009-08-26 20:21:42.344378 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1...@127.0.0.1:5062) Running State Change CS_ROUTING
2009-08-26 20:21:42.345314 [DEBUG] switch_core_state_machine.c:484
(sofia/internal/1...@127.0.0.1:5062) State ROUTING
2009-08-26 20:21:42.345314 [DEBUG] mod_sofia.c:130
sofia/internal/1...@127.0.0.1:5062 SOFIA ROUTING
2009-08-26 20:21:42.345314 [DEBUG] switch_ivr_originate.c:63
(sofia/internal/1...@127.0.0.1:5062) State Change CS_ROUTING - CS_CONSUME_MEDIA
2009-08-26 20:21:42.345314 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1...@127.0.0.1:5062 [BREAK]
2009-08-26 20:21:42.345314 [DEBUG] switch_core_state_machine.c:484
(sofia/internal/1...@127.0.0.1:5062) State ROUTING going to sleep
2009-08-26 20:21:42.346255 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1...@127.0.0.1:5062) Running State Change CS_CONSUME_MEDIA
2009-08-26 20:21:42.346255 [DEBUG] switch_core_state_machine.c:503
(sofia/internal/1...@127.0.0.1:5062) State CONSUME_MEDIA
2009-08-26 20:21:42.349173 [DEBUG] sofia.c:3289 Channel
sofia/internal/1...@127.0.0.1:5062 entering state [calling][0]
2009-08-26 20:21:42.350242
Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
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/1...@209.71.254.33 parsing [ROUTING-PEER_01]
continue=false
Dialplan: sofia/internal/1...@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 ${sip_h_X-ROUTE} should be PEER_01
for success?
Here is my default.xml:
?xml version=1.0 encoding=utf-8?
!-- http://wiki.freeswitch.org/wiki/Dialplan_XML --
include
context name=default
extension name=LOOKUP_ROUTE
condition field=destination_number expression=(\d+)$
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
action application=export data=sip_h_X-ROUTE=LOOKUP/
action application=bridge
data=sofia/internal/${destination_numb...@127.0.0.1:5062/
action application=set
data=ROUTE_GW=${sip_redirect_contact_user_0}/
action application=set data=AREA=${sip_redirect_contact_user_0}/
action application=transfer data=${destination_number} XML
ROUTING/
/condition
/extension
/context
context name=ROUTING
extension name=PEER_01
condition field=${sip_h_X-ROUTE} expression=PEER_01
action application=set data=hangup_after_bridge=true/
action application=set
data=continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION/
action application=set data=PEER=1.1.1.1/
action application=bridge
data=sofia/external/${destination_numb...@1.1.1.1/
action application=set data=PEER=2.2.2.2/
action application=bridge
data=sofia/external/${destination_numb...@2.2.2.2/
action application=set data=PEER=3.3.3.3/
action application=bridge
data=sofia/external/${destination_numb...@3.3.3.3/
/condition
/extension
/context
/include
--
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
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/1...@209.71.254.33 parsing [ROUTING-PEER_01]
continue=false
Dialplan: sofia/internal/1...@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:
--
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] How to Configure SIP DID to IVR
Оригинално писмо От: Michael Jerris Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR До: freeswitch-users@lists.freeswitch.org Изпратено на: Сряда, 2008, Юли 2 19:24:22 EEST ^ seems like an invalid regex. is that literally what you have there or you have some number? Mike On Jul 2, 2008, at 12:16 PM, Hristo Benev wrote: Hi, I'm new to FS and trying to configure DID only configuration. Here is the setup: PSTN Cisco AS(realIP/maybe multiple ones in production) FS(realIP) Cisco box is configured to send SIP to IP (real IP nor 192.168.x.x type) and I do not have much control over it. No authentication is needed. I'm using FS 1.0.0 What I need to configure to send incoming PSTN calls to demo IVR What I've changed? Created cisco.xml file in /conf/directory/default / / / -- Added to /conf/dialplan/default.xml - -- When I call DID it just rings. If I connect to FS with SoftPhone on extension and I dial DID. I was able to get this configuration working with Asterisk(but had some sound quality issues and wanted to try something else) so there is no HW problem. Where is my misconfiguration(hopefully just this)? Thanks ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Yes there is an actual number that I do not wanted to disclose. I have some progress now call are accepted by FS, but something is wrong after dialplan_hunt() is executed it hangs up. Thanks ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to Configure SIP DID to IVR
Here is the output:
---
2008-07-02 13:48:47 [NOTICE] switch_channel.c:533 switch_channel_set_name() New
Channel sofia/cisco/CallingNumber@CIscoIP
[c0d8586f-f6b9-4108-8676-c49e66f32e6d]
2008-07-02 13:48:47 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing
CAllingNumber-DIDNumber@cisco
2008-07-02 13:49:12 [ERR] sofia_glue.c:450 sofia_glue_ext_address_lookup() Stun
Failed! stun.freeswitch.org:3478 [Timeout]
2008-07-02 13:49:12 [NOTICE] mod_sofia.c:386 sofia_answer_channel() Hangup
sofia/cisco/CallingNumber@CiscoIP [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
2008-07-02 13:49:12 [NOTICE] switch_core_session.c:753
switch_core_session_thread() Session 1 (sofia/cisco/CallingNumber@CicoIP)
Ended
2008-07-02 13:49:12 [NOTICE] switch_core_session.c:755
switch_core_session_thread() Close Channel
sofia/cisco/CallingNumber@CiscoIP [CS_HANGUP]
---
CallinfNumber is the number I call from
CiscoIP is IP of Cisco AS
DIDNumber is DID I have
Thanks
I'm doing something wrong, but what?
Again Here are the files
/conf/sip_profiles/cisco.xml (just copied external.xml and changed sip port)
---
!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --
profile name=cisco
!-- This profile is only for cisco --
gateways
X-PRE-PROCESS cmd=include data=cisco/*.xml/
/gateways
aliases
alias name=cisco/
/aliases
domains
domain name=$${domain} parse=true/
/domains
settings
param name=debug value=5/
param name=sip-trace value=no/
param name=rfc2833-pt value=101/
param name=sip-port value=5060/
param name=dialplan value=XML/
param name=context value=cisco/
param name=dtmf-duration value=100/
param name=codec-prefs value=$${outbound_codec_prefs}/
param name=hold-music value=$${hold_music}/
param name=use-rtp-timer value=true/
param name=rtp-timer-name value=soft/
param name=manage-presence value=false/
param name=aggressive-nat-detection value=true/
param name=inbound-codec-negotiation value=generous/
param name=nonce-ttl value=60/
param name=auth-calls value=false/
param name=rtp-timeout-sec value=1800/
param name=rtp-ip value=$${local_ip_v4}/
param name=sip-ip value=$${local_ip_v4}/
param name=ext-rtp-ip value=$${external_rtp_ip}/
param name=ext-sip-ip value=$${external_sip_ip}/
param name=rtp-timeout-sec value=300/
param name=rtp-hold-timeout-sec value=1800/
/settings
/profile
--
/conf/dialpaln/cisco.xml
-
!-- http://wiki.freeswitch.org/wiki/Dialplan_XML --
include
context name=cisco
extension name=cisco1
condition field=network_addr expression=^xxx\.xxx\.xxx\.xxx$/
condition field=destination_number expression=^$
action application=answer/
action application=sleep data=2000/
action application=ivr data=demo_ivr/
/condition
/extension
extension name=cisco2
condition field=network_addr expression=^xxx\.xxx\.xxx\.xxx$/
condition field=destination_number expression=^$
action application=answer/
action application=sleep data=2000/
action application=ivr data=demo_ivr/
/condition
/extension
extension name=cisco3
condition field=network_addr expression=^xxx\.xxx\.xxx\.xxx$/
condition field=destination_number expression=^xxx$
action application=answer/
action application=sleep data=2000/
action application=ivr data=demo_ivr/
/condition
/extension
extension name=cisco4
condition field=network_addr expression=^xxx\.xxx\.xxx\.xxx$/
condition field=destination_number expression=^xxx$
action application=answer/
action application=sleep data=2000/
action application=ivr data=demo_ivr/
/condition
/extension
/context
/include
--
Sensitive data is obfuscated
Оригинално писмо
От: Michael Jerris
Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Сряда, 2008, Юли 2 20:22:31 EEST
Most likely its not actually matching the extension or it runs out of
actions to perform, can you post the full debug logs from the console?
Mike
On Jul 2, 2008, at 1:14 PM, Hristo Benev wrote:
Оригинално писмо
От: Michael Jerris
Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Сряда, 2008, Юли 2 19:24:22 EEST
^ seems like an invalid regex. is that literally what
you have there or you have some number?
Mike
On Jul 2, 2008, at 12:16 PM, Hristo Benev wrote:
Hi,
I'm new to FS and trying
Re: [Freeswitch-users] How to Configure SIP DID to IVR (Maybe fixed)
Strange I changed regex to DID not ^DID and it worked?!
Оригинално писмо
От: Hristo Benev
Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Сряда, 2008, Юли 2 21:00:59 EEST
Here is the output:
---
2008-07-02 13:48:47 [NOTICE] switch_channel.c:533 switch_channel_set_name()
New Channel sofia/cisco/@ [c0d8586f-f6b9-4108-8676-c49e66f32e6d]
2008-07-02 13:48:47 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing
-@cisco
2008-07-02 13:49:12 [ERR] sofia_glue.c:450 sofia_glue_ext_address_lookup()
Stun Failed! stun.freeswitch.org:3478 [Timeout]
2008-07-02 13:49:12 [NOTICE] mod_sofia.c:386 sofia_answer_channel() Hangup
sofia/cisco/@ [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
2008-07-02 13:49:12 [NOTICE] switch_core_session.c:753
switch_core_session_thread() Session 1 (sofia/cisco/@) Ended
2008-07-02 13:49:12 [NOTICE] switch_core_session.c:755
switch_core_session_thread() Close Channel sofia/cisco/@ [CS_HANGUP]
---
CallinfNumber is the number I call from
CiscoIP is IP of Cisco AS
DIDNumber is DID I have
Thanks
I'm doing something wrong, but what?
Again Here are the files
/conf/sip_profiles/cisco.xml (just copied external.xml and changed sip port)
---
--
/conf/dialpaln/cisco.xml
-
--
Sensitive data is obfuscated
Оригинално писмо
От: Michael Jerris
Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Сряда, 2008, Юли 2 20:22:31 EEST
Most likely its not actually matching the extension or it runs out of
actions to perform, can you post the full debug logs from the console?
Mike
On Jul 2, 2008, at 1:14 PM, Hristo Benev wrote:
Оригинално писмо
От: Michael Jerris
Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Сряда, 2008, Юли 2 19:24:22 EEST
^ seems like an invalid regex. is that literally what
you have there or you have some number?
Mike
On Jul 2, 2008, at 12:16 PM, Hristo Benev wrote:
Hi,
I'm new to FS and trying to configure DID only configuration.
Here is the setup:
PSTN Cisco AS(realIP/maybe multiple ones in production)
FS(realIP)
Cisco box is configured to send SIP to IP (real IP nor 192.168.x.x
type) and I do not have much control over it. No authentication is
needed.
I'm using FS 1.0.0
What I need to configure to send incoming PSTN calls to demo IVR
What I've changed?
Created cisco.xml file in /conf/directory/default
/
/
/
--
Added to /conf/dialplan/default.xml
-
--
When I call DID it just rings.
If I connect to FS with SoftPhone on extension and I dial DID.
I was able to get this configuration working with Asterisk(but had
some sound quality issues and wanted to try something else) so there
is no HW problem.
Where is my misconfiguration(hopefully just this)?
Thanks
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Yes there is an actual number that I do not wanted to disclose.
I have some progress now call are accepted by FS, but something is
wrong after dialplan_hunt() is executed it hangs up.
Thanks
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