sr-users January 2010

sr-users@lists.kamailio.org

73 participants
127 discussions

[Kamailio-Users] Does uac_auth work for you?
by Andrey Utkin 13 Jan '10

13 Jan '10

Hi all. Please share your experience. If you ever tried authenticating outgoing calls with Kamailio, did it work for you? Which devices or servers you know do/don't accept such authentication? Do you use it in some tricky way to get it working?

4 3

[Kamailio-Users] about kamailio with sctp
by mustafa samara 13 Jan '10

13 Jan '10

Hello i do not know how to try sctp, imake kmailio 1.5 with sctp=1 could you help me? thnaks -- Eng.Mustafa Al-Samara

3 2

Kamailio (OpenSER) v3.0.0 Released
by Daniel-Constantin Mierla 12 Jan '10

12 Jan '10

Hi everybody, this ML was forgotten yesterday to be cc-ed ...for the sake of archiving, here it is the announcement. After ten months since previous major release 1.5.0, I am glad to announce that Kamailio (OpenSER) v3.0.0 is out. It is a special release, being the first based on SIP Router project. Along with new things developed during the past months, you can blend features and modules provided by previous versions of Kamailio (OpenSER) and SIP Express Router (SER) in same configuration file. Given these, 3.0.0 offers an impressive number of new features and improvements, read release notes at: http://www.kamailio.org/mos/view/Kamailio-OpenSER-v3.0.0-Release-Notes Wiki page with more details about what is new: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x The tarball with sources: http://www.kamailio.org/pub/kamailio/latest/src/ Binary packages for several distributions: http://www.kamailio.org/pub/kamailio/latest/bin/ http://www.kamailio.org/pub/kamailio/latest/packages/ Modules' documentation: - http://www.kamailio.org/docs/modules/3.0.x/ Cookbooks and more wiki docs: - http://www.kamailio.org/dokuwiki/ Migration guidelines: - http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 From Kamailio project point of view, this is the first major milestone accomplished within SIP Router project, proving its viability. The work done so far, to offer the features brought out by this release, together with the development teams working together, make the future of the SIP Router project very solid. I would like to thank to everybody that contributed to get SIP Router project at this stage and I am eager to get to next major release from it source tree -- the work will continue in order to offer a more tight integration of Kamailio (OpenSER) and SIP Express Router (SER) projects, of course, along with development of new features. Cheers, Daniel

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[Kamailio-Users] Checking the destination IP after dispatcher
by Geoffrey Mina 12 Jan '10

12 Jan '10

What pseudo variable would i check after running ds_next_dst() to check the IP we are about to forward the INVITE to? Basically I want to have a check for $si == [something] to ensure I am not about to send the INVITE to the same UI which just requested it. While I have your attention, I may as well see if anyone has a better solution for what I am trying to accomplish: I am doing this because asterisk can't handle a hairpin scenario and I need to generate two distinct calls in my network, so I figure i'll just forward the request to another server in the network to deal with this. INVITE from PSTN --> Kamailio --> Asterisk#1 [Asterisk#1 needs to create a new session, but stay on-net] --> INVITE from Asterisk#1 to Kamailio --> [Dispatcher] --> INVITE from Kamailio to Asterisk#2 This would basically create 3 legs of the call, which is what I want PSTN --> Asterisk#1 --> Asterisk#2 I would prefer if Asterisk#1 could just INVITE to LocalHost, but that doesn't seem to be allowed. Not sure if that's a limitation of SIP or Asterisk... Thanks!

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[Kamailio-Users] job offer: VoIP developer, team Bucharest
by Henning Westerholt 12 Jan '10

12 Jan '10

Job offer: Interested in working with one of the biggest Voice over IP networks in europe? We're hiring and have one additional VoIP developer position open in the team in Bucharest, Romania at the moment. For more informations (requirements, how to apply..) please refer to the opening on the job web page. Some facts about our VoIP system can be found at e.g. [1], please feel free to contact me per mail as well. http://cariere.1and1.ro/xml/JobsDetail?job_id=2142 Best regards, Henning Westerholt [1] http://www.kamailio.org/events/2009-linuxtag/presentation.pdf -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

1 0

[Kamailio-Users] This has probably been asked a thousand times before...
by Robert B 12 Jan '10

12 Jan '10

Regardless, despite my Googlage, I can't seem to find what I am looking for. I've looked at the Wiki and the documentation. While I do have a programming background, I still am having a hard time grasping some of these tutorials. Many of them seem very specific and not at all relevant to what I am trying to do. I feel like I'm left without a good starting point. My goal is to have Kamailio setup as an ITSP would. That is, I have three sipxecs PBX systems here, and rather than have each one configured with umpteen SIP trunks that I'll need to manage, I want to have (for the time being) just one -- that one would be my personal Kamailio server. When a user makes a call, it first goes to my Kamailio server which then distribute the calls to my ITSPs (like Bandwidth.com, voip.ms, etc). I want the ACLs to be setup based on IP addresses and not user registrations. I want nothing to do with the media or RTP streams. First step would be termination (outbound). The obvious next step would be setting it up for origination (inbound). I've downloaded and installed the SIREMIS web UI for OpenSER 1.4.x, but as I said my lack of understanding of the terminology's context prohibits me from really exploring anything. Some areas look familiar, others seem redundant, etc. The reason I started with the web UI was because it's obviously easier and it also produces output that is syntactically correct for my own examination and learning purposes. What would the appropriate terminology for this kind of setup be? Can anyone point out a Wiki article which I may have missed? To me, this seems like the most basic of basic configurations and a seemingly obvious launching point for anyone getting into the software... Thanks! -- Robert

3 2

[Kamailio-Users] Kamailio (OpenSER) v3.0.0 Released
by Daniel-Constantin Mierla 11 Jan '10

11 Jan '10

Hi everybody, after ten months since previous major release 1.5.0, I am glad to announce that Kamailio (OpenSER) v3.0.0 is out. It is a special release, being the first based on SIP Router project. Along with new things developed during the past months, you can blend features and modules provided by previous versions of Kamailio (OpenSER) and SIP Express Router (SER) in same configuration file. Given these, 3.0.0 offers an impressive number of new features and improvements, read release notes at: http://www.kamailio.org/mos/view/Kamailio-OpenSER-v3.0.0-Release-Notes Wiki page with more details about what is new: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x The tarball with sources: http://www.kamailio.org/pub/kamailio/latest/src/ Binary packages for several distributions: http://www.kamailio.org/pub/kamailio/latest/bin/ http://www.kamailio.org/pub/kamailio/latest/packages/ Modules' documentation: - http://www.kamailio.org/docs/modules/3.0.x/ Cookbooks and more wiki docs: - http://www.kamailio.org/dokuwiki/ Migration guidelines: - http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 From Kamailio project point of view, this is the first major milestone accomplished within SIP Router project, proving its viability. The work done so far, to offer the features brought out by this release, together with the development teams working together, make the future of the SIP Router project very solid. I would like to thank to everybody that contributed to get SIP Router project at this stage and I am eager to get to next major release from it source tree -- the work will continue in order to offer a more tight integration of Kamailio (OpenSER) and SIP Express Router (SER) projects, of course, along with development of new features. Cheers, Daniel

3 2

[Kamailio-Users] preparing Kamailio 3.0.0 release
by Daniel-Constantin Mierla 11 Jan '10

11 Jan '10

Hello, as announced previously, today will package the kamailio 3.0.0 release. Please commit any change to kamailio_3.0 branch before 14:00UTC to avoid overlappings. If you are aware of new bugs, just drop an email to devel mailing list. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 1

Re: [Kamailio-Users] Users Digest, Vol 56, Issue 19
by mustafa samara 11 Jan '10

11 Jan '10

thank you very much i will see kamailio 3.0 but is there any voip soft phone which support sctp, please? thanks

2 1

Re: [Serusers] configure SER to work as a gateway
by "Martínez García, Rebeca" 11 Jan '10

11 Jan '10

Hi all, Thanks Rupert for your answer, I realise that my explanation was not very clear but the scenario I am working into is a little bit complex. Finally, I have decided to change my configuration and make the client sends its REGISTER message to SER and then it will forward the same message to my Server. However, the Server carries out the authentication process (I can not change it), so SER has to save the location contact after the client is authenticated in the Server. My function for the REGISTER message in SER is the following one: route[REGISTER] { t_newtran(); # send the register to the proxy rewritehostport("open-ims.test:4060"); t_on_reply("REGISTER_reply"); #t_on_failure("REGISTER_failure"); if (!t_relay()) { sl_send_reply("500","Error forwarding to P-CSCF"); break; } } onreply_route[REGISTER_reply] { if (t_check_status("401")){ log(-1, "A response from the P-CSCF!"); break; } if (t_check_status("200")){ log(-1, "User registered in the IMS Core!"); save_contacts("location"); break; } } But SER does not start. It shows this error: cng@cng:~$ sudo /etc/init.d/ser start -P /var/run/ser/ser.pid Starting ser: ser 0(2545) parse error (147,19-20): Command cannot be used in the block I have checked where is the error and it complains about the "save_contacts("location"))", but I do not know why I can not use it inside the onreply_route. Thanks in advance, Rebeca Martinez -----Mensaje original----- De: rupert.organ(a)bt.com [mailto:rupert.organ@bt.com] Enviado el: vie 08/01/2010 15:03 Para: Martínez García, Rebeca Asunto: RE: [Serusers] configure SER to work as a gateway Hi Rebeca, I think I understand what you are trying to do, albeit you describe it in a strange way. SER is designed to be a Registrar / SIP Proxy. It is not designed to proxy REGISTER messages. However I had to interface it to a SIP Application Server that wanted to see the REGISTER messages. However SER was my REGISTRAR and performed my security / authentication. Therefore I filter on the REGISTER messages using if (method=="REGISTER") { after doing the auth challenge in the ser.cfg, and after doing an eNum lookup and some user agent field checks I simply t_replicate the REGISTER. i.e. t_replicate("rebeca.node.es.com", "5060"); or if you do not use DNS t_replicate("10.1.x.y", "5060"); Note this replication does not include the auth challenge....but it satisfys my SIP app server functionality nicely, Also you can overwrite the useragent field if you wish... e.g. if (subst_uri("/;userAgent=rupert")) { subst('/^User-Agent:.*$/User-Agent: Rebeca/ig'); Good luck Rupert ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of "Martínez García, Rebeca" Sent: 08 January 2010 11:52 To: serusers(a)lists.iptel.org Cc: "Huertas García, Victor" Subject: [Serusers] configure SER to work as a gateway Hello all, Solved the installation problem I had, now I am trying to configure SER in a certain way. My configuration is the following one: Client ------------------------- SER ---------------------------- Server/Proxy 192.168.x.x 192.168.x.y 10.1.x.x 10.1.x.y Unlike the normal case where SER is used as the Registrar, my client (in a private LAN) must register in a Server/Proxy and SER has to act as a SIP gateway. However, at the same time, SER has to register locally this client (for internal calls). So, put in other words, SER should forward the REGISTER message (directed from the client to the Server/Proxy) and save locally the location of that client. Once registered the user, SER should act as a transparent proxy. But in the practice, it does nothing if I put that the client registers in the Server/Proxy. The messages reach SER but it does not forward them, nor it registers locally the clients. So I tried configuring the client to register in SER instead of the Server/Proxy and then SER started to work. Is a listening problem? What I have to include to the ser.cfg so SER handles also messages which are not directed to it but to the Server/Proxy? I hope somebody could help me. Thanks in advance, Rebeca Martinez

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sr-users January 2010