Hello Daniel,
I am not programmer, but I want place request if kaamilio devs can complete it .
Thanks
Slava.
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Tuesday, 15 November, 2016 03:00:19
Subject: Re: [SR-Users] msilo
Hello,
for me it is fine to add a new parameter to m_store() that takes the body content.
Are you asking for more details of how can be done?
Cheers,
Daniel
On 13/11/16 07:08, Slava Bendersky wrote:
Hello Everyone,
I asking question/request to add improvement for msilo module where will have ability set $avp(i:body) in mod params. That will allow use it in m_store() by specifying body message $avp. In my case registration is on B2BUA so kamailio pass through/proxy to destination. In order to check user status online/offline kamailio wait for reply from B2BUA on MESSAGE status if it return error 503 that mean user offline and m_store(0 should store message in db, but the issue that error 503 is in final stage and MESSAGE body is not available any more. The improvement will allow specify body $var in m_store() so new way will be m_store($tu, $avp(body)).
Any help thank you.
Slava.
_______________________________________________
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
I have tried $Rp before sending this to mailing list, but with $Rp I
have different values for example 188, 6847, 3163, 9175 for every call
?!
Don't know where are they come from, because in URI I see :6060, and
using $rp give my 6060 for examples with $Rp.
Only have problem with UAC Panasonic-MPR12-V004.41009/VSIPGW-V3.0000
with doesn't send $rp in URI.
Greetings
I managed to create about 1900 concurrent calls using a single Kamailio and
RTPProxy server. But after this number RTPProxy returns 0 and the following
error is shown in the Kamailio log files:
incorrect port 0 in reply from rtp proxy
What is the problem here? Also number of file descriptors that
RTPProxy can use are set to a million.
Hello
My point is to log INVITEs in my Network. My SIP Network work on port
6060.
Im am using HEP to do that, and hep clients that are listening on port
range 5060-6066 to detect some SIP attack to port 5060 and others.
But when I have attack to port 5060 I don't want to insert that INVITE
to my "good traffic" table, but place it to "fraud" table.
Some devices like Panasonic PBX send INVITE to port 6060 so it is "good
traffic" but there is no port 6060 in URI, so I can't detect it in right
way, because $dp or $rp are 5060 by default.
This kamailio work as capture server as "promiscuous_on", so I can't use
any "force_rport" etc. because I am only listening. Tcpdump for that SIP
session show me that client send traffic to 6060, but I can't get that
information from INVITE header.
Greetings
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Daniel Pocock <daniel(a)pocock.pro>
I have changed the following tm parameters in order to have an timeout
INVITE equal to 20 sec (20000 ms).
# ----- tm params -----
# auto-discard branches from previous serial forking leg
modparam("tm", "failure_reply_mode", 0)
# default retransmission timeout: 20sec
modparam("tm", "fr_timer", 20000)
# default invite retransmission timeout after 1xx: 120sec
modparam("tm", "fr_inv_timer", 120000)
However, the timeout is ever 3 min (180 sec.)
In accord to tm module the timeout is defined only by "r_timer"
Then, I don't know where is the problem.
Thanks a lot for your help.
--
Diogenes
Hello Everyone,
I asking question/request to add improvement for msilo module where will have ability set $avp(i:body) in mod params. That will allow use it in m_store() by specifying body message $avp. In my case registration is on B2BUA so kamailio pass through/proxy to destination. In order to check user status online/offline kamailio wait for reply from B2BUA on MESSAGE status if it return error 503 that mean user offline and m_store(0 should store message in db, but the issue that error 503 is in final stage and MESSAGE body is not available any more. The improvement will allow specify body $var in m_store() so new way will be m_store($tu, $avp(body)).
Any help thank you.
Slava.
Hello Daniel,
I resolved all configuration issues and right now call goes both directions include rtp.
Again huge thank you for help.
Slava.
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "volga629" <volga629(a)skillsearch.ca>
Cc: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Thursday, 10 November, 2016 12:24:57
Subject: Re: [SR-Users] BYE dispatcher
Hello,
this logic is definitely wrong -- FreeSwitch can send also a request, it means that you send it back to it.
Only the initial request of a dialog should be routed with rules like dispatcher/load balancer/least cost routing/... The requests within dialog should be routed based on loose routing.
Of course, one can think of exceptions, but then you should be fully aware of what kind of routing you do.
Cheers,
Daniel
On 10/11/16 16:25, Slava Bendersky wrote:
Hello Daniel,
My setup is proxy all requests to freeswitch via dispatcher.
Slava.
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "volga629" <volga629(a)skillsearch.ca> , "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Thursday, 10 November, 2016 04:56:53
Subject: Re: [SR-Users] BYE dispatcher
Hello,
as I said before, the registrations have little to do with calls in sip, unless there is gruu in use.
Cheers,
Daniel
On 09/11/16 18:07, Slava Bendersky wrote:
BQ_BEGIN
Hello Everyone,
I cleared registrations and tried again and issue still present.
Client reply with 481.
IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638)
client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610
E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0
Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp
From: "Test Extension" <sip:4300@sip.company.tld> ;tag=SXt3DQQ90a0Dj
To: < sip:4300@ client_pub_ip :49383>;tag=719973534
Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb
CSeq: 99019404 BYE
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
Slava.
BQ_END
--
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
Hi guys,
I'm using captagent on my all kamalios and freeswitches to capture and
send sip to homer server. That works very well.
But I would like to enable, on homer's kamailio, the module SipTrace
(http://www.kamailio.org/docs/modules/4.4.x/modules/siptrace.html) and
save captured data into mysql as well.
I already set it up using this mysql structure :
http://www.kamailio.org/docs/db-tables/kamailio-db-devel.html#idp1279328
Almost everything works except the column "DIRECTION". Each row is saved
as "IN" direction.
Do you guys think that can be solved ? If so, can you pleas direct me
HOW ...
Thx
Jan F.
Hello
I have some UAC like Panasonic PBX, that send traffic to port 6060 that
I am listening on,
but that port isn't included to R-URI.
I can see that only at tcpdump, or sip_trace from sip_capture module.
Kamailio variables like $dp or $rp have default value of 5060.
Thank You,
