Hi all,
I am still stuck with the ACK message not being forwarded by the
originating PCSCF. Any advice would be great.
Thanks,
Serhat
On 24 October 2016 at 21:00, Serhat Guler <srtguler(a)gmail.com> wrote:
> Hi Daniel,
>
> I am using only record_route() without any parameters. I do not have a
> proper computer atm to draw the network diagram, but I can tell you shortly
> about the network setup.
>
> I have only enabled websockets for the pcscf to allow ws and wss
> connections. In that case there is a ws connection that uses UDP protocol.
> This is the ACK to complete the session setup.
>
> the sipml5 client is configured as follows:
> WebSocket Server URL: ws://192.168.0.11:880
> SIP outbound Proxy URL: udp://192.168.0.11:4060
>
> Mercuro IMS client: uses UDP port as well: 4060
>
> The call is made from sipml5 client. The Mercuro phone rings, and when I
> reply the call, 200 OK is sent to sipml5 webrtc client, but the ACK from
> sipml5 doesn't pass the PCSCF as I explained in the previous message.
>
> A part of PCSCF cfg file:
>
> # Check for Subsequent requests:
> if (has_totag()) {
> # sequential request withing a dialog should
> # take the path determined by record-routing
> if (loose_route()) {
> if ($route_uri =~ "sip:mo@.*") {
> setflag(FLT_MO);
> }
> if(!isdsturiset()) {
> handle_ruri_alias();
> }
> # RTP-Relay, if necessary
> route(RTPPROXY);
> t_relay();
> } else {
> if ( is_method("ACK") ) {
> if ( t_check_trans() ) {
> # no loose-route, but stateful ACK;
> # must be an ACK after a 487
> # or e.g. 404 from upstream server
> t_relay();
> exit;
> } else {
> xlog("L_INFO", "ACK without matching transaction ...
> ignore and discard!!!!!\n");
> # ACK without matching transaction ... ignore and
> discard
> exit;
> }
> }
> sl_send_reply("404","Not here");
> }
> exit;
> }
>
> Cheers,
> Serhat
>
>
>
> On 24 October 2016 at 20:18, Daniel-Constantin Mierla <miconda(a)gmail.com>
> wrote:
>
>> Hello,
>>
>> I haven't noticed the log files, it's ok.
>>
>> From the Route header, I see that there is a proxy that uses WS:
>>
>> Route: <sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy
>> 1nCMEI1mR0RztrB;did=e82.0c3>
>> That is the address of the next hop and typically a proxy doesn't use
>> websocket connection to another proxy. Can you show a diagram with the sip
>> server nodes in your network and what protocols are used between them?
>>
>> Are you simply use record_route() function, or some other function or
>> different parameters to it?
>>
>> Cheers,
>> Daniel
>>
>>
>> On 24/10/16 12:18, Serhat Guler wrote:
>>
>> Hi Daniel,
>>
>> Thanks for your reply. I actually attached a log file with debug level 3,
>> consisting ACK related messages. If you would like to see more logs, I'll
>> send a new log file in the evening.
>>
>> Cheers,
>> Serhat
>>
>> On 24 October 2016 at 12:13, Daniel-Constantin Mierla <miconda(a)gmail.com>
>> wrote:
>>
>>> Hello,
>>>
>>> can you get all the log messages for ACK but with debug=3 in the
>>> kamailio.cfg?
>>>
>>> Cheers,
>>> Daniel
>>>
>>> On 23/10/16 22:04, Serhat Guler wrote:
>>>
>>> Hello,
>>>
>>> I finally managed to place a call from sipml5 webrtc client to Mercuro
>>> IMS client. The phone rings, and when I answer it sends 200 OK to the
>>> sipml5 where as sipml5 send back an ACK message which never passes the
>>> originating PCSCF. The PCSCF says:
>>>
>>> 8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS
>>> connection (id: 0) for WebSocket could not be found
>>> 8(3640) ERROR: <core> [msg_translator.c:1947]:
>>> build_req_buf_from_sip_req(): could not create Via header
>>> 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building
>>> failed
>>>
>>> I doubt that the WebSocket connection is closed, cause when I terminate
>>> the call from Mercuro client a bye request is being sent to the sipml5.
>>>
>>> The ACK package:
>>>
>>> ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2.
>>> Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9
>>> hG4bKvuly7bmxnN4aqM4zZTIS;rport
>>> From: "Bob"<sip:bob@net1.test>;tag=GxzKy1nCMEI1mR0RztrB
>>> To: <sip:alice@net1.test>;tag=18823
>>> Contact: "Bob"<sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=no;click2c
>>> all=no;transport=ws>;+g.oma.sip-im;language="en,fr"
>>> Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6
>>> CSeq: 3887 ACK
>>> Content-Length:
>>> Route: <sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp>
>>> Max-Forwards: 69
>>> Route: <sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy
>>> 1nCMEI1mR0RztrB;did=e82.0c3>
>>> Route: <sip:mo@192.168.0.11:4060;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0Rz
>>> trB;did=e82.0c3>
>>> Route: <sip:mo@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di
>>> d=e82.f062>
>>> Route: <sip:mt@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di
>>> d=e82.f062>
>>> Route: <sip:mt@192.168.0.11:4060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di
>>> d=e82.1c3>
>>> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
>>> Organization: Doubango Telecom
>>>
>>> Have been thinking for quite a while, but couldn't really find a reason
>>> why it wouldn't add the v,a header. A debug 3 level log file is also
>>> attached.
>>>
>>> Thanks in advance,
>>> Serhat
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>> --
>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com
>>>
>>> _______________________________________________ SIP Express Router
>>> (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users(a)lists.sip-router.org http://lists.sip-router.org/cg
>>> i-bin/mailman/listinfo/sr-users
>>
>> --
>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com
>>
>>
>
Hello,
I was wondering if it is possible to do a failover in digest
authentication. For eg: I get two possible valid password values. Can I do
a pv_proxy_authenticate in such a way where if authentication fails, I do a
pv_proxy_autenticate again on the different value to check if that succeeds?
Is it even allowed in digest authentication mechanism. Just getting
opinions.
Thanks for any help.
- Jayesh
Hello Daniel,
No that exactly the one,
Slava.
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Friday, 28 October, 2016 07:37:31
Subject: Re: [SR-Users] fetch contact
Hello,
the contact address is one for attributes in $ulc(...), iirc, it is the =>addr -- see the wiki for pv cookbook for more details. Or are you looking for something else?
Cheers,
Daniel
On 27/10/16 21:43, Slava Bendersky wrote:
Hello Everyone,
How to fetch contact from location table ?
I tried look reg_fetch_contacts(), but I don't specific $ulc for contact.
Thanks
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello!
Given a Diversion header like the following:
Diversion:"Foo
Bar"<sip:+9876543211@127.0.0.101;user=phone>;privacy=off;answered-count=2;re
ason=deflection;counter=1;answered,"_
somewhere"<sip:+9876543212@127.0.0.101;user=phone>;privacy=off;answered-coun
t=1;reason=deflection;counter=1;answered
Kamailio 4.4.3 (and probably earlier versions) emit error messages
complaining about the comma character. According to RFC 6044 multiple
comma-separated diversions in a single Diversion-header is allowed.
Full INVITE as an example:
INVITE sip:+1234567890@127.0.0.100:5060;user=phone;transport=tcp SIP/2.0
Via:SIP/2.0/TCP
127.0.0.101;branch=z9hG4bKBroadWorks.-1od4eg8-127.0.0.100V5060-0-82844591-13
54681939-1477634439005-
From:<sip:+9876543210@127.0.0.101;user=phone>;tag=1354681939-1477634439005-
To:<sip:+1234567890@127.0.0.100:5060;user=phone>
Call-ID:BW0800390052810161562691307@127.0.0.101
CSeq:82844591 INVITE
Contact:<sip:127.0.0.101:5060;transport=tcp>
P-Asserted-Identity:<sip:+9876543210@127.0.0.101;user=phone>
Privacy:none
Diversion:"Foo
Bar"<sip:+9876543211@127.0.0.101;user=phone>;privacy=off;answered-count=2;re
ason=deflection;counter=1;answered,"_
somewhere"<sip:+9876543212@127.0.0.101;user=phone>;privacy=off;answered-coun
t=1;reason=deflection;counter=1;answered
Supported:100rel
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
X-BroadWorks-DNC:network-address="sip:+9876543210@127.0.0.101;user=phone"
Accept:application/dtmf-relay,application/media_control+xml,application/sdp,
application/x-broadworks-call-center+xml,multipart/mixed
Max-Forwards:10
Content-Type:application/sdp
Content-Disposition:session;handling=required
Content-Length:267
v=0
o=BroadWorks 100442303 1 IN IP4 127.0.0.101
s=-
c=IN IP4 127.0.0.102
t=0 0
m=audio 29778 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
Example log messages:
Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core>
[parser/parse_addr_spec.c:488]: parse_to_param(): ERROR parse_to_param :
invalid character ',' in status 21:
<<;privacy=off;answered-count=2;reason=deflection;counter=1;answered>>
Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core>
[parser/parse_addr_spec.c:254]: parse_to_param(): ERROR: parse_to_param
:unexpected char ["] in status 21:
<<;privacy=off;answered-count=2;reason=deflection;counter=1;answered,>> .
Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core>
[parser/parse_addr_spec.c:906]: parse_addr_spec(): ERROR: parse_to: invalid
To - unexpected end of header in state 21
Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core>
[parser/parse_diversion.c:69]: parse_diversion_header():
ERROR:parse_diversion_header: bad diversion header
Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core>
[lvalue.c:345]: lval_pvar_assign(): non existing right pvar
Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core>
[lvalue.c:405]: lval_assign(): assignment failed at pos: (562,21-562,35)
The code on line 562 is $avp(div_user) = $(di{uri.user});.
Can this be handled in any way by Kamailio without changing the parser?
Best regards,
Björn Bylander
my logfile /var/log/kamailio/kamailio.log is empty.
1)
/etc/rsyslog.d/kamailio.cfg exists with the entry:
log_facility=LOG_LOCAL0
2)
/etc/rsyslog.con has the entry:
#
# don't log messages with LOG_LOCAL0 in /var/log/syslog anymore
*.*;auth,authpriv.none,local0.none -/var/log/syslog
#
# log messages with LOG_LOCAL0 in /var/log/kamailio.log
local0.* -/var/log/kamailio.log
#
local0.* -/var/log/kamailio/kamailio/kamailio.log
Alternatively I edited /etc/rsyslog.d/kamailio.conf that should be used by
rsylog.conf
3)
/etc/rsyslog.d/kamailio.conf exists with the entry:
local0.* -/var/log/kamailio/kamailio.log
I am working with ubuntu on an amazon instance.
Why is the logfile empty?
Thanks Herman
i used the cnxcc module its successfully terminated the call. but if you
want i can help you.
On Thu, Oct 27, 2016 at 12:58 AM, <sr-users-request(a)lists.sip-router.org>
wrote:
> Send sr-users mailing list submissions to
> sr-users(a)lists.sip-router.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> or, via email, send a message with subject or body 'help' to
> sr-users-request(a)lists.sip-router.org
>
> You can reach the person managing the list at
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>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of sr-users digest..."
>
>
> Today's Topics:
>
> 1. Re: cnxcc doesn't terminate calls (Igor Potjevlesch)
> 2. logfile is empty (Hermann Norpois)
> 3. Re: Out of memory in UB 210: OOM killed process 12261
> (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB
> (Jurijs Ivolga)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 26 Oct 2016 12:48:29 +0200
> From: "Igor Potjevlesch" <igor.potjevlesch(a)gmail.com>
> To: "'Kamailio \(SER\) - Users Mailing List'"
> <sr-users(a)lists.sip-router.org>
> Subject: Re: [SR-Users] cnxcc doesn't terminate calls
> Message-ID: <009c01d22f76$7d0c9a30$7725ce90$(a)gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi,
>
>
>
> Yep, the dialog is created. I put the dialog into DB and see it.
>
> Nothing appears in the logs when the timer is reached.
>
>
>
> I succeed to do the same behaviour with dialog module only and
> dlg_set_timeout. Looks to do why I want.
>
>
>
> I'm a bit frustrating about cnxcc because I really don't understand why
> nothing happen after max_time, but, anyway, it works with dialog only.
>
>
>
> Regards,
>
>
>
> Igor.
>
>
>
> De : sr-users [mailto:sr-users-bounces@lists.sip-router.org] De la part
> de Arsen
> Envoyé : mercredi 26 octobre 2016 11:15
> À : Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org>
> Objet : Re: [SR-Users] cnxcc doesn't terminate calls
>
>
>
> Hi Igor,
>
>
>
> Make sure that you create a dialog for this call and $dlg_var(key) is
> available (This pseudo-variable will be available only for subsequential
> requests after doing loose_route().)
>
>
>
> Also check kamailio logs maybe you try to terminate call for wrong
> customer id.
>
>
>
> On Wed, Oct 26, 2016 at 10:52 AM, Igor Potjevlesch <
> igor.potjevlesch(a)gmail.com <mailto:igor.potjevlesch@gmail.com> > wrote:
>
> Hi,
>
>
>
> I'm testing cnxcc module in order to make a simple call duration
> limitation.
>
> My understanding is to use the function like this :
>
>
>
> $dlg_var(customer_id) = "customer-" + $fU;
>
> $var(max_time) = 5;
>
>
>
> if (!cnxcc_set_max_time("$dlg_var(customer_id)",
> "$var(max_time)"))
>
>
>
> Unfortunately, after 5 seconds, nothing happen. I don't even see an error
> from dialog module.
>
>
>
> Anyone has a better understanding of the way the module runs? Thank you!
>
>
>
> Regards,
>
>
>
> Igor.
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
>
>
>
>
> --
>
> Regards,
> Arsen.
>
>
Hi,
I'm testing cnxcc module in order to make a simple call duration limitation.
My understanding is to use the function like this :
$dlg_var(customer_id) = "customer-" + $fU;
$var(max_time) = 5;
if (!cnxcc_set_max_time("$dlg_var(customer_id)",
"$var(max_time)"))
Unfortunately, after 5 seconds, nothing happen. I don't even see an error
from dialog module.
Anyone has a better understanding of the way the module runs? Thank you!
Regards,
Igor.
Hello,
I try to use app_perl and cnxcc module.
As soon as I load the module like this:
# --------- app_perl params ----
modparam("app_perl", "filename",
"/usr/local/etc/kamailio/AAA/aaa-script.pl")
modparam("app_perl", "modpath", "/usr/local/lib64/kamailio/perl/")
The Via header is modified for each calls:
/usr/local/sbin/kamailio[21064]: INFO: <core> [parser/msg_parser.c:355]:
parse_headers(): ERROR: bad header field [Via:ASIP/2.0/UDP 91.]
/usr/local/sbin/kamailio[21064]: ERROR: <core> [parser/parse_via.c:2048]:
parse_via(): ERROR: parse_via: bad char <A> on state 100
What could explain this behaviour?
Regards,
Igor.
