sr-users April 2011

sr-users@lists.kamailio.org

64 participants
105 discussions

Notifying RLS when XML documents are updated
by Peter Dunkley 18 Apr '11

18 Apr '11

Hello, I am using Kamailio RLS and integrated XCAP Server. When I add a contact with my client it updates my XCAP documents but there is no additional SIP messaging between the client and Kamailio. I have been assured (by the client author) that this is the correct behaviour. However, having looked at both the XCAP Server and RLS module code I cannot see anything that will either inform RLS that a document has been updated or a timer to check for that. Have I missed something obvious? Thanks, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

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perlvdb error
by Roman Yeryomin 18 Apr '11

18 Apr '11

Hi! I'm trying to use perlvdb module but seems that it's not working. Seems that needed functions are not exported correctly: kamailio[11525]: ERROR: <core> [db.c:79]: module db_perlvdb does not export db_use_table function Looking at perlvdb.c one can see that module exports it's functions as "perlvdb", not "db_perlvdb". But changing "perlvdb" to "db_perlvdb" results in segfault when running: kamailio[31336]: segfault at 1 ip b7365e3a sp bfa82fb0 error 4 in perlvdb.so[b7362000+6000] Do I misunderstand something completely or there is a bug? Anyone actually tried using this module or it just sits there forgotten? Regards, Roman

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Probing mode howto?
by Spinov Evgeniy 18 Apr '11

18 Apr '11

Hello, all. Is there any possibility to set one of dispatcher destinations to probing mode manually? I can set it to active/inactive, however cannot set it to probing via MI. This is really required as, for instance, when asterisk reload takes like 1 minute to reload ( huge DB ), I want to set node to inactive first to do not make call attempts, and after reload set it to probing. Thank you.

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loose_route security
by Eric Hiller 18 Apr '11

18 Apr '11

As I look and play with loose_route functionality it seems that by simply placing a route: proxyip;lr header in my invite I can bypass any and all security otherwise built into the configuration. Is this the way everyone has it? I have been unable to find any configuration examples online that show how to secure/restrict access to loose_route? Ideas or suggestions? Thanks, Eric

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Fwd: [RAI] SIPit 28 summary
by Klaus Darilion 17 Apr '11

17 Apr '11

-------- Original-Nachricht -------- Betreff: [RAI] SIPit 28 summary Datum: Fri, 15 Apr 2011 10:47:08 -0500 Von: Robert Sparks <rjsparks(a)nostrum.com> An: rai(a)ietf.org Also available at <https://www.sipit.net/SIPit28_summary> RjS -------- SIPit 28 was hosted by Digium in Huntsville, Alabama, USA the week of Arpil 11-15, 2010. There were 54 attendees from 19 companies visiting from 10 countries. We had 40 distinct implementations. The roles represented (some implementations act in more than one role) 34 endpoints 6 proxy/registrars/b2bua/sbcs 1 dedicated event server Implementations using each transport for SIP messages: UDP 98% TCP 95% TLS 68% server-auth, 50% mutual-auth SCTP 10% DTLS 0% 68% of the implementations present supported IPv6. There was one RFC4474 Identity implementation present. For DNS we had support for: Full RFC3263 : 53% SRV only : 33% A records only : 15% no DNS support : 0% Support for various items in the endpoints: 76% replaces 38% 5389stun 26% turn 24% 3489stun 24% ice 21% sip/stun multiplexing 21% outbound 21% gruu 18% path 18% join 18% diversion 6% history-info (there were no implementations of 4244bis) 3% service-route 3% sigcomp Support for various items in the proxy/registrars: 50% path 50% outbound 33% sip/stun multiplexing 20% diversion 18% gruu 17% history-info 17% service-route The endpoints and B2BUAs implemented these methods: 100% INVITE, CANCEL, ACK, BYE 94% REGISTER 91% OPTIONS 91% REFER 82% NOTIFY 82% INFO 79% UPDATE 71% SUBSCRIBE 68% PRACK 47% PUBLISH 29% MESSAGE 94% of the implementations sent RTP from the port advertised for reception (symmetric-rtp). One of the implementations required the other party to use symmetric-rtp. 88% of the UAs present both sent RTCP and paid attention to RTCP they received. 56% of the endpoints present supported SRTP using sdes. There was one dtls-srtp implementations at this SIPit. 67% of the endpoints supported multipart/MIME. There were no implementations present with S/MIME support. 48% of the implementations present followed RFC6026 (corrections to the INVITE transaction) 68% followed RFC4320 (corrections to the non-INVITE transaction) There were 5 SIP Event Server implementations (not counting endpoints that support events only for refer). There were 9 SIP Event Client implementations (not counting endpoints that support events only for refer). These event packages were supported: Server Client 2 3 presence 1 0 presence.winfo 1 6 message-summary 2 5 dialog 0 2 reg 2 1 conference 1 0 xcap-diff 0 1 kpml These packages were supported with the eventlist extension: Server Client 1 0 presence Five of the proxies present still rely only on max-forwards. There was one implementation of fork-loop-fix, but no implementations of max-breadth. Multiparty tests (Thanks to Eoin McLeod for contributing notes for several tests) The Forking test was very well attended. We had the majority of the participants at the multiparty table. Many teams took away good new information from watching their implementation react to multiple merges and to Via stacks with transports switching between IPv4 and IPv6. Forcing multiple 200 Oks was more difficult than usual given the network equipment at hand. We've created a set of standing automated tests that will allow endpoints and proxies to test that scenario at their leisure. The TLS multiparty leveraged DNS to reduce configuration churn. We ensured a full mesh of connections between participating elements. There were some issues uncovered with the values used in Route/Record-Route not matching the configured certificates at an element. We brainstormed a configuration for the next event that will efficiently expose whether elements are correctly opening a new connection (rather than reusing an existing connection) when a new request with a resource belonging to an authority not matching what had been authenticated on the existing connection arrived. The SRTP tests showed a high level of interoperability (all with SDES keying), including hold/resume and transfer scenarios. There was a successful test of a call with media running long enough to increment the ROC, followed by a hold/resume. Several participants were trying variants of "best-effort/opportunistic" encryption with lower levels of success. These elements tested providing offers with both RTP/AVP and RTP/SAVP media-lines describing the same logical media channel, and with multiple RTP/AVP media lines decorated with different crypto attributes. The STUN/TURN/ICE tests had very high levels of interoperability. The test scenario involved elements in two natted networks sharing the same address space and elements in the public address space (including a separate TURN server for each of the private network islands). We had a mix of elements that supported relay allocation and elements that presented only host/reflexive candidates. A few elements assumed that all of the m-lines in an offer had to present ICE candidates. SDP with a mix of ICE enabled m-lines and m-lines without candidates caused ICE failures. The Outbound test showed several successful flows being created and maintained. A mesh of edge-proxies and proxy/registrars were set up and endpoints established flows through several combinations. There was good UA support for the Flow-Timer, but there were implementations present that didn't randomize the keepalive value. There was not much testing of recovery from failed flows at this event - we'll focus on that at SIPit 29. We held a long multiparty session focusing on exchanging video with constrained bandwidth or impaired (latent or lossy) networks. There was a high level of basic interoperability and reasonable reaction to mid-call shifts in network conditions, but all parties involved came away with changes they wanted to make in their implementations. The Spiral test presented requests to the participating UAs containing a mix of UDP, TCP, and TLS over IPv4 and IPv6 values in large Via and Record-Route stacks. Correct operation of subsequent in-dialog requests in both directions was tested, discovering a few implementation problems, and some invalid assumptions about how dialog-stateful proxies might need to behave, but most scenarios worked well. We followed the usual Spiral test with a "fuzz-ball" test utilizing DNS SRV to randomly distribute the propagation of messages through all of the proxies present, hopping off to the UAs with a low probability, resulting in an easy-to-configure test that exercised a full-mesh of connections between proxies and presented the UAs with very diverse (and large) Via and Record-Route header field values. We exercised the RFC5393 proxy-doom scenario, again leveraging the SRV load-leveling capabilities to simplify configuration. We had messages popping out to the UAs from each level of the tree, giving them a very large-scale merge test. _______________________________________________ RAI mailing list RAI(a)ietf.org https://www.ietf.org/mailman/listinfo/rai

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Meeting in San Francisco
by Daniel-Constantin Mierla 14 Apr '11

14 Apr '11

Hello, I will spend few days in San Francisco area, during May 6-10. If someone wants to meet and discuss about the project, VoIP & stuff, drop me an email. In case there are many people leaving close to SF city, then I can even try to arrange a group meeting, for a dinner or so, to map faces to names and have good drinks -- this decision will be announced later based on feedback, however suggestions about good meeting places are welcome from anyone. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [SR-Users] Problems with siremis custom charts
by Daniel-Constantin Mierla 14 Apr '11

14 Apr '11

Do not send me message privately upon answers on the mailing list - keep the mailing list always cc-ed otherwise the messages will be ignored. On 4/14/11 3:02 PM, Camila Troncoso wrote: > > Daniel, > > I want a bars graph and i´m using another table, one that I create. > The graph displays well with the default configuration (the one that > the other graph of siremis have) but I don´t want an area graph. I > search around siremis files and find a file named charts-lib.php. I > suppose this is the script that creates the graphs , but in this file > are only two type of graphs describe near line 130 ( area and > line_dot). I don’t know if this is why I can´t change the graph type. > yes, so far these are the ones supported by siremis for building custom graphs. The type, one of them, is taken from the xml config files describing the graphs. The library for making graphs supports other types, as you could see in the usrloc charts, but that is using a different php code base than the other grpahs. If you want to enhance it with other types of graphs, that would be great and if you send a working patch then we can include it in the next release. Cheers, Daniel > > Regards, > > Camila > > *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com > <mailto:miconda@gmail.com>] > *Sent:* jueves, 14 de abril de 2011 6:16 > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Cc:* Camila Troncoso > *Subject:* Re: [SR-Users] Problems with siremis custom charts > > Hello, > > On 4/13/11 10:56 PM, Camila Troncoso wrote: > > Hi, > > I´m having problems with siremis charts. I create a new graph and all > is working fine, but I want to use another type of graph ( the default > graphs of siremis uses type=”area”) and when I change the graph type > to any other thing it shows me the same graph but only the area under > the curve isn´t paint , but the curve doesn´t change. > > Please some help with this issue… > > what type of graph you try to use? have you checked the database > statistics table records, do they have different values for this > graph? If not, check if you write the desired values from the kamailio > config file there. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

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How I can turn off 100 trying reply
by Derrick Ding 14 Apr '11

14 Apr '11

Hi, All I wonder how I can make Kamailio not to send "100 Trying" msg when receive INVITE request, by modifying kamailio.cfg or source code. Thanks Derrick

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Problems with siremis custom charts
by Camila Troncoso 14 Apr '11

14 Apr '11

Hi, I´m having problems with siremis charts. I create a new graph and all is working fine, but I want to use another type of graph ( the default graphs of siremis uses type=”area”) and when I change the graph type to any other thing it shows me the same graph but only the area under the curve isn´t paint , but the curve doesn´t change. Please some help with this issue… Regards, Camila T. *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

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Problems with db_text
by Sascha Daniels 14 Apr '11

14 Apr '11

Hi together, I am trying to make a setup with kamailio and freeswitch, that can run with all supported database types. The basic setup is from this howto: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc With mysql everything is (more or less) fine. Creating db_text databases does not work because /opt/kamailio-3.1/share/kamailio/dbtext/kamailio/uacreg is missing and kamdbctl gives up. Just created an empty file, so kamdbctl can finish its work. The subscribers database is created (works in a more simple setup). Now I am getting this error when starting kamailio: http://pastebin.com/vsiUNdCM Version of table: pua:6 Why does the table version not exist? It was created by "kamdbctl create". I am a bit lost with this debug output. Greets Sascha -- AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied --> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister Montabaur B14998 Bücher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de

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sr-users April 2011