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
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
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
>>
>>
>
> 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......."
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
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/
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
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
Hello all,
I'm pretty new to the SIP world and appreciate if somebody can help me out.
In the SerWeb login page I noticed a link "Have-My-Domain!", and it looks like that the result would be having something similar to email forwarding service. Suppose that I have my own domain (e.g. mydomain.com), and I have added an SRV record to it and pointed it to iptel's sip server. Then following the link I was able to create a new username (e.g. user01). Now based on what I've understood, whenever somebody calls the sip uri user01(a)mydomain.com my soft phone that is registered with the above account should ring (my soft phone is using user01 for username, mydomain.com for server and iptel.org for registerar and it seems that it is registered without any problem).
My questions:
1. What prevents others from impersonating a new username in my domain ? It seems that if somebody knows that mydomain.com has an SRV record pointed to iptel.org, can create another username (user02(a)mydomain.com) and use it without my knowledge, what is worse is that my DNS provider doesn't have any logging options for DNS requests, so I'll never find out about it.
2. In my account in iptel ( user01(a)mydomain.com ), there is another sip uri ( something like 123456(a)mydomain.com ), which I guess is supposed to be used by somebody who uses a device without alphabets (for example if somebody wants to call me via sipbroker gateways), I've already tried *478 - 123456 ( *478 is the code for iptel), but it doesn't work and it goes to voice mail, how can I use sipbroker to call user01(a)mydomain.com ?
Thanks a lot in advance and sorry for the long message !
Someone
