sr-users November 2008

sr-users@lists.kamailio.org

97 participants
171 discussions

[Serusers] DNS failover
by Nuno Ribeiro 05 Nov '08

05 Nov '08

Hi all, I have a scenario where I was testing DNS failover and this feature seems to fail. I will try to explain the scenario: I'm trying to establish a call between two user Agents, where one of the users has services associated (trigger to ASs) to the session establishment signaling (INVITE). So when SER makes the DNS query receives 2 ip addresses for this AS. The sip message will be correctly forwarded to one of this two entries and if in case of failure it will try to send the message to the next entry (second ip address). But when we are triggering to the AS we will add a route header and this header is not created in the second case, i.e. after DNS failover as the SIP INVITE will be created based in the SIP message received and not in the SIP message sent in the first place. So anyone has any thoughts about this question and a solution to it? Thanks in any advance. Best Regards -- Nuno Ribeiro

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[Kamailio-Users] SER/Asterisk interworking mailing list.
by Alex Balashov 05 Nov '08

05 Nov '08

Greetings, As a developer and consultant who spends considerable time on projects involving the fusion of Asterisk and products derived from the SER ecosystem (OpenSER, Kamailio, OpenSIPS, the new SIP-Router), I have found that there is a great volume of interest in this topic on the mailing lists associated with all communities involved, but a comparative lack of focus that results in duplicated effort and lack of specialised response. This is mainly due, I think, to the fact that detailed Asterisk experience - while common - is not a prerequisite for working with the SER products, while for Asterisk people SER can often be a next step in scalability and VoIP service delivery platform enhancement that they are just getting into. And so on. There's pollution in the respective discursive spaces; a lot of Asterisk people posting to the SER lists ask a lot of Asterisk-specific questions in addition to any they may have about SER which can be construed as potentially off-topic by some members, and the opposite is true on the Asterisk lists when detailed, involved discussion about SER occurs. We need to capture that discussion that exists at the overlap and is specifically concerned with making these two systems work together, requiring somewhat detailed and esoteric understanding of both and a community of user support and knowledge that focuses on both of these conceptual and product universes. Toward that end, I am hosting a new mailing list with this succinct purpose, if slightly unwieldy name, and encourage all interested to join. It is called 'SER-Asterisk-Interwork' and can be accessed for subscription here: http://lists.evaristesys.com/mailman/listinfo/ser-asterisk-interwork The archives are available here: http://lists.evaristesys.com/pipermail/ser-asterisk-interwork/ You can post to the list at: ser-asterisk-interwork(a)lists.evaristesys.com It's the same GNU Mailman stuff you are already used to. While it could be argued that this cross-product discussion is valuable to retain in both communities, I think there is considerable benefit to creating a specialised mailing list that focuses specifically on this integration path and the unique interoperation and configuration issues it creates. I think it would be good to get some of this discussion off of the SER and Asterisk-specific mailing lists where it has somewhat marginal relevance at times and refocus it. If you agree and are interested in this topic, you are invited to join the list. One last note: The SER/OpenSER community has been in a state of flux recently, with OpenSER undergoing a name change to Kamailio and subsequently seeing a fork. The incumbent Kamailio project is now in the process of merging with the original SER project. The choice of nomenclature for list is not meant to imply an endorsement of or affinity for the IPTel SER project per se. It is just that right now it serves the aim of terseness to use a common denominator, to refer to this family of projects as the "SER ecosystem." Whether you are a SER, OpenSER, Kamailio, or OpenSIPS user, you are part of that "SER ecosystem." That is why the list is named what it is. Thank you, -- Alex -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] Load userblacklist table to cache w/o restarting?
by Pablo Bertuzzi 05 Nov '08

05 Nov '08

Hello forum, I was wondering whether there is a fifo command or workaround to load the userblacklist table into memory without restarting. I have a large carrierroute table and it takes about 20 secs to start. best regards, PB

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Re: [Kamailio-Users] rewriteuser with AVP
by Samuel Muller 05 Nov '08

05 Nov '08

Hello Ravi , i'm using radiator but not freeradius, actually my users are stored into an user flat file. here you can have an example of user : 0123452000(a)sip.720.fr User-Password = "2000" User-Name = "0123452000", Sip-User-ID = "0123452000", Sip-User-Realm = "sip.720.fr", Sip-URI-User = "0123452000(a)sip.720.fr", Sip-Rpid = "0123452000", Sip-Group = "full", Sip-AVP = "asserted_id:0123452000" and concerning services attributes, you have to declare them like this : => in kamailio.cfg (or openser.cfg, ser.cfg, opensips.cfg) : modparam("acc|auth_radius", "service_type", 15) modparam("avp_radius", "caller_service_type", 30) modparam("avp_radius", "callee_service_type", 31) if you're using peering module : modparam("peering", "verify_destination_service_type", 21) modparam("peering", "verify_source_service_type", 22) => in the radius dictionary (for radiusclient-ng, and the dictionary used by your radius server) ### Service-Type Values ### VALUE Service-Type Call-Check 10 # RFC2865, uri_radius VALUE Service-Type Group-Check 12 # Proprietary, group_radius VALUE Service-Type Sip-Session 15 # Schulzrinne, acc/auth radius VALUE Service-Type Sip-Verify-Destination 20 # Proprietary, peering VALUE Service-Type Sip-Verify-Source 21 # Proprietary, peering VALUE Service-Type Sip-Caller-AVPs 30 # Proprietary, avp_radius VALUE Service-Type Sip-Callee-AVPs 31 # Proprietary, avp_radius => then, in your radius server config, you must add one handler per service-type example for radiator, just a base config based on flat user file: <Handler Service-Type=Sip-Verify-Destination> RewriteUsername s/^sip:(.*)/$1/ <AuthBy FILE> NoCheckPassword Filename %D/user.peer NoDefault </AuthBy> </Handler> I can't help you actually for a radius DB, 'cause i'm working on it. cheers, .Sam. -- Samuel MULLER Ingénieur Reseaux & Telecom 720 DEGRES +33 (0)663 128 505 sml(a)720.fr On Wed, Nov 5, 2008 at 1:28 PM, LetMeKnow < sunkara.raviprakash.feb14(a)gmail.com> wrote: > Hello Samuel, > Can you check the radius services attributes > > is you using users.conf or radius databases. > > can see this url > http://www.kamailio.org/docs/openser-radius-1.0.x.html#freeradius_users > > > > Thanks &Regards > Ravi Prakash Sunkara > VoIP Architect & JAVA-SIP Developer > +91-9999882776 > > > 2008/11/5 Samuel Muller <sml(a)720.fr> > >> Hello all, >> >> I've a little question : >> I got a username in an AVP by avp_load_radius, and I would rewrite the uri >> with it. >> >> The objectives is to replace the r-uri by an ipbx uri, then forward the >> call to it by a new branch (to not loose the phone number requested). >> in this case : 0123452000 is behind a spa9000. The user 0123452000 has an >> AVP called "ipbx", that i'm using to do some groups and permissions (plus >> the rewrite). >> >> actually, it does not work, i tried many ways : >> >> # load radius attributes of the callee >> if (!avp_load_radius("callee")) >> { >> xlog("L_INFO","-> user unknown in radius usr db : $ru"); >> route(15); # route PSTN >> } >> >> # verify the AVPs we got (caller and callee) >> xlog("L_INFO", " -AVP------------------------------------- "); >> avp_print(); >> xlog("L_INFO", " ----------------------------------------- "); >> >> # callee is behind an ipbx (avp ipbx) ? >> if (is_avp_set("$avp(s:callee_ipbx)")) >> { >> xlog("L_INFO", "-> callee is behind an ipbx : >> $avp(s:callee_ipbx)"); >> setflag(14); # flag IPBX >> #subst_user('/$rU$/$avp(s:callee_ipbx)/'); >> #rewriteuri("sip:$avp(s:callee_ipbx)@$rd"**); >> rewriteuser($avp(s:callee_ipbx)); >> } >> >> # callee in usrloc ? >> if (lookup("location")) >> { >> append_hf("P-hint: usrloc applied\r\n"); >> xlog("L_INFO","-> registered user called : $tu"); >> route(7); # route FORWARD >> } >> else >> { >> xlog("L_INFO","-> 480: local user not found in usrloc : >> $tu"); >> sl_send_reply("480","Temporarily Unavailable"); >> drop; >> } >> >> and in the debugs log : >> >> Nov 5 13:04:51 ser0 kamailio[2930]: >> -ROUTE--INBOUND-------------------------- >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: rc_auth >> Success >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: AVP >> 'callee_asserted_id'/0='0123452000'/0 has been added >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: AVP >> 'callee_ipbx'/0='spa9000'/0 has been added >> Nov 5 13:04:51 ser0 kamailio[2930]: >> -AVP------------------------------------- >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> p=0x7fb3d7e5cfe8, flags=0x0003 >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Iname=<callee_ipbx> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Ival_str=<spa9000 / 7> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> p=0x7fb3d7e5cf88, flags=0x0003 >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Iname=<callee_asserted_id> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Ival_str=<0123452000 / 10> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> p=0x7fb3d7e5ced8, flags=0x0003 >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Iname=<caller_asserted_id> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Ival_str=<0123451011 / 10> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> p=0x7fb3d7e5ce38, flags=0x0003 >> Nov 5 13:04:51 ser0 kamailio[2930]: >> ----------------------------------------- >> Nov 5 13:04:51 ser0 kamailio[2930]: -> callee is behind an ipbx : spa9000 >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:registrar:lookup: '$avp( >> s(a)sip.720.fr' Not found in usrloc >> Nov 5 13:04:51 ser0 kamailio[2930]: -> 480: local user not found in >> usrloc : sip:0123452000@sip.720.fr <sip%3A0123452000(a)sip.720.fr> >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:parse_headers: >> flags=ffffffffffffffff >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:check_via_address: params >> 77.246.81.162, 192.168.0.134, 0 >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:destroy_avp_list: destroying >> list 0x7fb3d7e5d040 >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:receive_msg: cleaning up >> >> >> -> But, if I did it by : >> >> if (uri =~ "^sip:012345200{1}") >> { >> rewriteuri("sip:spa9000@sip.720.fr<sip%3Aspa9000(a)sip.720.fr> >> "); >> } >> if (lookup("location")) >> { >> append_hf("P-hint: usrloc applied\r\n"); >> xlog("L_INFO","-> registered user called : $tu"); >> route(7); # route FORWARD >> } >> >> it works great, and the call is ok (the ipbx forward correctly the request >> the phone behind him). >> >> >> Anyone has an idea ? thanks in advance ! >> >> >> -- >> Samuel MULLER >> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >

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Re: [Kamailio-Users] Users Digest, Vol 42, Issue 11
by luzango mfupe 04 Nov '08

04 Nov '08

> 9. Re: the sip router project (Bogdan-Andrei Iancu) > > > Mmmh! guys, i am missing some points here, does this has anything to do > with OPENSIPS-YATE Collabo? a counter attack move of some sort??? > > > ------------------------------ > > Message: 9 > Date: Tue, 04 Nov 2008 17:24:15 +0200 > From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Subject: Re: [Kamailio-Users] the sip router project > To: Johansson Olle E <oej(a)edvina.net> > Cc: devel <devel(a)lists.kamailio.org>, serusers(a)lists.iptel.org, > sr-dev(a)lists.sip-router.org, users <users(a)lists.kamailio.org>, SER > developer mailing list <serdev(a)iptel.org>, > business(a)lists.kamailio.org > Message-ID: <4910691F.9020509(a)voice-system.ro> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi Olle, > > Thank you for your thoughts. On a first view, it looks interesting, but I'm > missing some points here (important points): > > 1) as OpenSER was forked from SER because different views (and the OpenSER > view proved to be a very popular and successful one), I wonder > why, Kamilio is getting back to SER? not sharing any more the OpenSER view > as claimed? because such merging will definitely have a great > impact on the dynamical and openness of the projects (like releases, > contributions, driving the project) > > 2) this major change of perspective (at least for kamilio) was a backstage > decision, kept secret from the community - shouldn't be in the > interest of the community to say if going back to the roots (merging into > SER) is something wanted or not? it somehow contradicts the self > existence of OpenSER, right? > > 3) the benefits you mentions are mainly optimization of the internal > project activities and not optimizations of the outcome - what the > project will deliver. And I guess this is the most important. We already > went though the experience of large devel community, frameworks, etc but > with no outcome for more than 2 years... > > >From my personal perspective, the new project looks more like SER > absorbing Kamilio (considering the sizes, the companies behind each > project, the resources, and the man-power behind each project). > > And at the moment I would like preserve the OpenSER vision and to have an > open source project (a standalone one), far away from the "control" of any > Big Brother ;) > > Regards, > Bogdan > > > > Johansson Olle E wrote: > > Congratulations to both projects! > > > > I think this is very beneificial for both communities as well as for > > all the companies out there using the two products. > > > > I am very happy with this result and hope that the third kid in the > > SER family - OpenSIPS - will consider joining the effort too. > > > > Best regards, > > /Olle > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > > > > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > End of Users Digest, Vol 42, Issue 11 > ************************************* > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Kamailio-Users] the sip router project
by Daniel-Constantin Mierla 04 Nov '08

04 Nov '08

Hello Everybody, We are pleased to announce to you the SIP Router project. It aims to build a solid open source SIP routing platform, based on collaboration of the SIP Express Router (SER) and Kamailio (OpenSER) teams. Developers of these two projects believe that an united and non-conflicting environment will bring many benefits to them, community members and companies: * bring together the developers and user communities of both projects * reduce maintenance overhead * avoid duplicated efforts in development * develop a core framework that is flexible, extensible and scalable * promote and build a solid open source SIP server project * ensure business credibility * make future forking undesirable, this harms everybody, affects credibility and business You are welcome to join! Visit the web site at: http://sip-router.org There is a meeting in Karlsruhe, Germany, on Nov 10, 2008, hosted by 1&1, where the developers and community members have the chance to discuss and tune the last aspects of the new project. We are looking to see many of you there: http://sip-router.org/index.php/meeting/ We hope this is a great news for you, thanks to the effort of main developers and management teams of the two projects. We invite you to join the new mailing list for further discussions: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev Daniel, Jiri Kamailio Management Team SER Management Team

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[Kamailio-Users] Unknown column 'from_uri' in 'field list'
by Kaeman Chris 04 Nov '08

04 Nov '08

Hello, modparam("acc", "db_extra", "from_uri=$fu; to_uri=$tu") added 2 new fields in the acc table. Logs giving following errors while making calls Nov 4 07:14:11 [4567] ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'from_uri' in 'field list' Nov 4 07:14:11 [4567] ERROR:core:db_do_insert: error while submitting query Nov 4 07:14:11 [4567] ERROR:acc:acc_db_request: failed to insert into database Is this error due to the added fields? Thanks, KChris Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

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[Kamailio-Users] SER config file adaptation to Kamailio ?
by Samuel Muller 04 Nov '08

04 Nov '08

Hello all, I tried during one week to adapt my old SER configuration file (ser.cfg, for SER 2.0.0), who works really great, to Kamailio v1.4.2. I used to change all the stuff needed (many functions, onsend route, ... disappeared or was replaced, ...). That's ok, concepts are the same - registration are ok, NAT (seems to) works with rtpproxy, and I'm using radius server (Radiator). MySQL is only here to store locations, domains, and trusted stuff like gateways. Everything looks fine (load kamalio : no pbm, REGISTER, OPTIONS methods and SIP ping ok, ...) Right now, RTP does not work, and no SIP phone (linksys spa9xx, thomson st2030, ...) can't call - there's no ringing, even if all the sip phones are registered and known by OpenSER ... It seems OpenSER can't send the SIP requests to the callee ??? Did anyone had the same problem and resolved it ? Or, eventually anyone does have some config sample config file (as you can have in SER examples stuff) ? I would like to let here a debug log of a call from a linksys spa942 to a sjphone but it's a bit enormous (debug log + config file) ... so i let it in attached file. the scenario is : 0123451011 (sjphone) calls 0123451012 (linksys spa9xx) the 2 UACs are on the same LAN, through the same router, with the same realm. there're registered and are stored into the location table in mySQL. but the problem is present in all the case we can have (different realms, different WAN connections, different SIP UACs ...) Thanks in advance ! -- Samuel MULLER

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[Kamailio-Users] Remote-Party-ID header
by ALAEDDINE abbech 04 Nov '08

04 Nov '08

hi all, Please, what is the role of Remote-Party-ID header? thanks

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[Kamailio-Users] PUA API send to certain IP
by Schumann Sebastian 04 Nov '08

04 Nov '08

Anca, All, I have the following scenario UE - OpenSER Proxy/Presence Server - OpenSER Registrar The proxy acts as proxy and presence server. If the UE sends a subscribe to user@registrar, it handles this subscription. If it publishes its state user@registrar it does as well. All registration/call control is proxied to the registrar. I try now to use the PUA module API to create PUBLISH on the Proxy/Presence Server. This works fine, the only issue is that it has the request-uri user@registrar, like UE generated subscriptions. This is actually not an issue, but wanted. The problem is that the proxy forwards the package to the registrar acc. R-URI. I want it sent to another IP. Is there the possibility to define the IP where the package should be sent to, i.e. presence server IP, to handle the packet there? Thanks! Best regards, Sebastian

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sr-users November 2008