sr-users

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4 participants
37456 discussions

[Serusers] cancelling forked requests + forcing TLS use
by Andrew T Gin 08 Mar '07

08 Mar '07

Hi I have two questions. 1. Receiver(a)1.1.1.1 and Receiver(a)2.2.2.2 register with ser. Client sends an INVITE to ser for Receiver. ser forks these INVITES in parallel to both Receiver(a)1.1.1.1 and Receiver(a)2.2.2.2. As both Receivers are online, they both reply with 200 OK. SER then relays BOTH 200 OK's back to the Client. My question: Shouldn't SER send CANCEL to one of the receivers, and relay only ONE 200 OK back to the client? How would I do this? It is not the client's responsibility to send a CANCEL to the SER to relay to one of the receivers, is this correct? This is because the client only sent one invite, therefore it should only receive one OK. ie should SER 'shield' the client from the knowledge of the existence of multiple contact addresses? 2. When I have tls_verify_client = 1 tls_require_client_certificate = 1 in my cfg file, ser still accepts tcp and udp connections, despite tls_require_client_certificate=1. I did a search, and found this way: if (proto != TLS) { sl_send_reply("403", "Forbidden"); exit; }; However SER is really still listening on UDP and TCP port 5060. Is there a way to configure SER such that it only listens and acknowledges TLS connections? Thanks Andrew

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[Users] CLI and GUI
by Michel Bensoussan 08 Mar '07

08 Mar '07

Hello Is there is CLI (Command Line Interface) that enable a user to read or eventually to change parameters at run time in OpenSER? Is there a GUI (web, php,...) that enable configuration? Thanks. Regards, Michel.

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[Serusers] Prague User Group Meeting Agenda
by Greger V. Teigre 08 Mar '07

08 Mar '07

Hi, Who's in charge of the agenda in Prague? (Or do we have somebody in charge? ;-) The presenters need to be allowed time to prepare for presentations etc. And we should decide on the length and the format of the meeting. I hope we can avoid a "death-by-powerpoint" (yes, I include openoffice in this category) style and rather have short intros and discussions :-) g-)

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[Users] pdt and IPv6
by Bernard Chan 08 Mar '07

08 Mar '07

Dear All, I am experimenting with OpenSER on an IPv6 environment. The pdt module appears not to support using an IPv6 address as destination domain (I guess that is true of source domain too, but I do not need that anyway). I tried to append a pdt row to MySQL using the documented OpenSER FIFO interface. The message I sent to the FIFO is similar to :pdt_add:openser_tmp_reply voip.mycompany.com 9 [::ffff:abcd:1234] However, when the row is created in MySQL pdt, the [:: is apparently missing. I guess that is because the FIFO parser mistook that as the FIFO :-separated message header or something similar? Testing with an IPv4 address instead as the destination domain, the row can be created properly, and prefix2domain() worked flawlessly in an IPv4 environment. I would like to use IPv4-mapped IPv6 address to forward to an Asterisk PABX over SIP+IPv4 as Asterisk has no official IPv6 release yet (this is another question, whether OpenSER will handle v4-mapped address properly). This is what I would like to test by setting this pdt rule. I'm using OpenSER 1.2.0-dev19-notls on i386/linux, by the way. Thanks for any invaluable insights. Regards, Bernard Chan.

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[Serusers] SEMS 0.10.0 rc2 released
by Stefan Sayer 07 Mar '07

07 Mar '07

Hi, We are happy to announce the release of SEMS 0.10.0 rc2. This release has a number of bugfixes and stability improvements, but also some interesting new application and component modules like PIN collect for scalable conference systems, a new flexible way to script SEMS in Python, SIP registrar client and UACi authentication, and a new method to handle jitter. Check it out at http://iptel.org/sems . From the Changelog - new Adaptive jitter buffer as alternative playout method Contributed by Andriy Pylypenko/Sippy Software - new PIN collect application with transfer to e.g. separate conference bridge - new SIP registrar client for registration at a SIP registrar - new UAC authentication component - new faster announcement application with memory caching for audio files - new pre call announcement method using REFER - stats server can be used for monitoring custom modules/applications - session specific parameters by default taken from unified session parameters header - signature configurable - install and make system updated - added documentation - some security bugfixes (namely fixing possible buffer overflows) - ...and a lot of other bug fixes Thanks go to Andriy and Maxim for the adaptive jitter buffer, Juha, Alexandr, Alex and all the others for bug reports and contributions. Regards Stefan for the semsdev team -- Stefan Sayer Media Services Development stefan.sayer(a)iptego.de www.iptego.de iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

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[Serusers] SIP SER
by Goh, Tee-Yong 07 Mar '07

07 Mar '07

Dear all, Would appreciate your advice from the ser-0.9.6 program, if save.c is the program in the SER that reply the 200 OK message for the REGISTER sends from the client? Thanks. Regards, Howard This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer

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[Users] [user] radius_is_user_in problem.
by Bodin Bruno 07 Mar '07

07 Mar '07

Hello, I try to use this code : if (!radius_is_user_in("From", "voip")) { sl_send_reply("403", "Forbidden"); exit; }; Radius server return this : Reply-Message = "Authorized" but in openser in debug mode there is : 0(12528) radius_is_user_in(): Failure Thank for your help.

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[Users] RE: Openser and Sip provider
by Gene Willingham 07 Mar '07

07 Mar '07

It is possible to have the SIP provider route calls to an IP address. This will alleviate the need to register with the provider. Using a Linksys device may not be the answer either. The Linksys device can register with openser, but there are limitations on the handling of inbound calls. The linksys device assumes that the TO Header is the account name. There is a way to route the inbound DNID using the dial plan. You can contact Linksys support, they should be able to tell you. I would if I remembered how. For asterisk we modified the code (very simple change) to allow Openser to route calls to asterisk and still use the users peer definition. The change is a very small change 3 lines of code, that works with both 1.2 and 1.4 branch of Asterisk. We can share this with you. Gene Willingham Telasip -----Original Message----- Message: 1 Date: Tue, 6 Mar 2007 14:32:56 -0600 From: "Scott Yagel" <syagel(a)packetcall.net> Subject: [Users] Openser and SIP provider To: <users(a)openser.org> Message-ID: <000901c7602e$9fefbf40$4b67a8c0@syagel> Content-Type: text/plain; charset="us-ascii" Hello, Am I correct in my assumption that I can't directly connect Openser to a ITSP due to Openser not providing registration to the ITSP? This seems to be what I read in the Openser documentation. This being the case, I have a Linksys SPA9000 pbx that I can use as a gateway to the ITSP, but attempts to route 10-digit numbers to the pbx get a response of 403 Forbidden from the pbx (the pbx has registered to the Openser OK). I see that many use an asterisk for a gateway, how do they get around this problem with the asterisk? Thanks, Scott Yagel PacketCall, Inc. syagel(a)packetcall.net

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[Serusers] Setup call forward by phone keyboard
by inge 07 Mar '07

07 Mar '07

Hi all, Is there a way to setup the call forward using the phone ? For example, proxy handle uri *73*xxxxxxxxxx#. Next is it possible to store "xxxxxxxxxx" into usr_preferences table like ? Thanks for your support

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[Users] WeSIP for commercial use
by Helmut Kuper 07 Mar '07

07 Mar '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I remember, that WeSIP isn't free for commercial use. So I does anyone knows anything about pricing model? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF7Xqt4tZeNddg3dwRAi8FAJ9RZBjIfkmgiLtPfd14214AI6hllQCgvEP0 8ykyRmMA0EsGOj+1h0tiznE= =oHku -----END PGP SIGNATURE-----

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