sr-users November 2016

sr-users@lists.kamailio.org

57 participants
123 discussions

Re: [SR-Users] msilo
by Slava Bendersky 15 Nov '16

15 Nov '16

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

1 0

How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 15 Nov '16

15 Nov '16

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

3 2

RTPProxy benchmark
by Gholamreza Sabery 15 Nov '16

15 Nov '16

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.

3 9

How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 15 Nov '16

15 Nov '16

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

2 1

[CFP] FOSDEM 2017, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 15 Nov '16

15 Nov '16

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2017 takes place 4-5 February 2017 in Brussels, Belgium. https://fosdem.org This email contains information about: - Real-Time communications dev-room and lounge, - speaking opportunities, - volunteering in the dev-room and lounge, - related events around FOSDEM, including the XMPP summit, - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities), - the Planet aggregation sites for RTC blogs Call for participation - Real Time Communications (RTC) ======================================================= The Real-Time dev-room and Real-Time lounge is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. The dev-room is a successor to the previous XMPP and telephony dev-rooms. We are looking for speakers for the dev-room and volunteers and participants for the tables in the Real-Time lounge. The dev-room is only on Saturday, 4 February 2017. The lounge will be present for both days. To discuss the dev-room and lounge, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc To be kept aware of major developments in Free RTC, without being on the discussion list, please join the Free-RTC Announce list: http://lists.freertc.org/mailman/listinfo/announce Speaking opportunities ---------------------- Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications dev-room: deadline 23:59 UTC on 17 November. Please use the Pentabarf system to submit a talk proposal for the dev-room. On the "General" tab, please look for the "Track" option and choose "Real-Time devroom". https://penta.fosdem.org/submission/FOSDEM17/ Other dev-rooms and lightning talks: some speakers may find their topic is in the scope of more than one dev-room. It is encouraged to apply to more than one dev-room and also consider proposing a lightning talk, but please be kind enough to tell us if you do this by filling out the notes in the form. You can find the full list of dev-rooms at https://www.fosdem.org/2017/schedule/tracks/ and apply for a lightning talk at https://fosdem.org/submit Main track: the deadline for main track presentations is 23:59 UTC 31 October. Leading developers in the Real-Time Communications field are encouraged to consider submitting a presentation to the main track at https://fosdem.org/submit First-time speaking? -------------------- FOSDEM dev-rooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the dev-room administrators personally if you would like to ask any questions about it. Submission guidelines --------------------- The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - the purpose of your talk - any other talk applications (dev-rooms, lightning talks, main track) - availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and dev-room themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the dev-room administrators. As the two previous dev-rooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. Volunteers needed ================= To make the dev-room and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 4 February - participation in the Real-Time lounge - helping attract sponsorship funds for the dev-room to pay for the Saturday night dinner and any other expenses - circulating this Call for Participation to other mailing lists See the mailing list discussion for more details about volunteering: https://lists.fsfe.org/pipermail/free-rtc/2016-October/000285.html Related events - XMPP and RTC summits ===================================== The XMPP Standards Foundation (XSF) has traditionally held a summit in the days before FOSDEM. There is discussion about a similar summit taking place on 2 and 3 February 2017 http://wiki.xmpp.org/web/Summit_21 - please join the mailing list for details: http://mail.jabber.org/mailman/listinfo/summit We are also considering a more general RTC or telephony summit, potentially in collaboration with the XMPP summit. Please join the Free-RTC mailing list and send an email if you would be interested in participating, sponsoring or hosting such an event. Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 3 February. On Saturday night, there are usually dinners associated with each of the dev-rooms. Most restaurants in Brussels are not so large so these dinners have space constraints and reservations are essential. Please subscribe to the Free-RTC mailing list for further details about the Saturday night dinner options and how you can register for a seat: https://lists.fsfe.org/mailman/listinfo/free-rtc Spread the word and discuss =========================== If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: All projects http://planet.freertc.org planet(a)freertc.org XMPP http://planet.jabber.org ralphm(a)ik.nu SIP http://planet.sip5060.net planet(a)sip5060.net (Español) http://planet.sip5060.net/es/ planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ======= For any private queries, contact us directly using the address fosdem-rtc-admin(a)freertc.org and for any other queries please ask on the Free-RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc The dev-room administration team: Saúl Ibarra Corretgé <saghul(a)gmail.com> Iain R. Learmonth <irl(a)debian.org> Ralph Meijer <ralphm(a)ik.nu> Daniel-Constantin Mierla <miconda(a)gmail.com> Daniel Pocock <daniel(a)pocock.pro>

2 1

INVITE timeout timer
by Diogenes Aviles 15 Nov '16

15 Nov '16

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

3 2

msilo
by Slava Bendersky 15 Nov '16

15 Nov '16

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.

2 1

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 14 Nov '16

14 Nov '16

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

2 1

Homer's kamailio config + enabling siptrace module
by Ján Füri 11 Nov '16

11 Nov '16

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.

2 3

How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 11 Nov '16

11 Nov '16

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,

3 2

← Newer
1
...
5
6
7
8
9
10
11
12
13
Older →

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users November 2016