sr-users July 2008

sr-users@lists.kamailio.org

147 participants
206 discussions

by Stephen Paterson

Don't worry about this, I've found the problem and it was nothing to do with SER - the reliable 1xx was being sent without a contact header so the PRACK was being sent with the same RURI as the initial INVITE. Nothing SER could do but fork the PRACK. All working now I've fixed the UAS. Cheers Steve -----Original Message----- From: Stephen Paterson Sent: 15 July 2008 10:21 To: 'Michal Matyska' Cc: serusers(a)iptel.org Subject: RE: [Serusers] PRACK on forked response Hi Michal, Thanks for the reply and sorry for the delay. Had me a nice relaxing long weekend in the country! Anyhow, I've not set up anything special in the config file. I just installed SER and ran it straight out of the box as it were. It worked fine for my purposes so I felt no need to tinker with the config. There is no mention of forking or lookup_contacts in the config. Diff shows the only difference between my config file and the simple example one that comes with the distribution is at the top in the global configuration parameters section where I set a listen address. All the routing logic is identical. Cheers Steve -----Original Message----- From: Michal Matyska [mailto:michal.matyska@iptel.org] Sent: 09 July 2008 15:11 To: Stephen Paterson Cc: serusers(a)iptel.org Subject: Re: [Serusers] PRACK on forked response Hi, SER does not care about PRACK... it is just another within-dialog request type, like BYE and many others. I think you have incorrect routing script logic, where you fork even the within dialog requests (using lookup_contacts) instead of just following the record route set. Michal Stephen Paterson píše v St 09. 07. 2008 v 14:01 +0100: > Hi all, > > I'm using SER purely for testing purposes just now. Currently testing > our handling of forked responses and hit upon the following situation: > > My outgoing INVITE is forked to two UASs (also our own SIP endpoint), > one of which sends a reliable 180 and the other an unreliable 180. The > UAC sends a PRACK on the dialog ID of the reliable response but SER > forwards this response on to both UASs the result being the UAS that > sent the unreliable 180 responds to the PRACK with a 481 (the PRACK > contains a To-tag that part defines a dialog unknown to that UAS). I > confess I can't find anything describing this scenario in any RFC but > it strikes me as incorrect behaviour on SERs part. > > Does SER support PRACK? If not then no matter, I can live with it - it > doesn't affect the outcome of any call. If it does, does anyone have > any thoughts on this? > > Cheers > > Steve > > Registered Address Lakeside, Bramley Road, Mount Farm, Milton Keynes, > MK1 1PT, UK > Registration No: 1397386 (Wales) > > P Please consider the environment and don't print this e-mail unless > you really need to > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers Registered Address Lakeside, Bramley Road, Mount Farm, Milton Keynes, MK1 1PT, UK Registration No: 1397386 (Wales) P Please consider the environment and don't print this e-mail unless you really need to

16 years, 4 months

[OpenSER-Users] SIMPLE-XMPP Developer Workshop 2008

by Daniel-Constantin Mierla

Hello, for those interested in SIP/SIMPLE - XMPP(Jabber/GTalk) interoperability, there is going to be a workshop in Paris for developers in the first week of September. Agenda for openser is to identify a reliable xmpp library that follows latest XMPP specs and offers secure communication (TLS) which can be used to update and enhance the simple-xmpp gateway implementation. In addition, it is a good opportunity to deploy and test it against other implementations. If you have extra ideas for openser sip-xmpp, please write me or, better, join the event and help get it done. Detailed description of the event follows. Best regards, Daniel SIMPLE-XMPP Developer Workshop 2008 SIP/SIMPLE and XMPP share a lot of concepts but they are different in many aspects. Being nowadays the leading open protocols for voice, video, instant messaging and presence, the interconnection between them creates an unified communication environment for users in both sides. The workshop aims to bring together people with large expertise in both protocols, interested in development, testing and deployment of SIP/SIMPLE-XMPP solutions. With a permanent focus on innovation, the participants cover open source projects to private enterprises. You can join the event for free. Date: September 2-5, 2008 Place: INRIA, Paris, France Main organizers: - Philippe Sultan, INRIA - contributor to the XMPP/Jingle support in Asterisk, member of XMPP Standards Foundation - Olle E. Johansson, Edvina - main SIP developer of Asterisk - Daniel-Constantin Mierla, Asipto - co-founder Openser, developer of XMPP/Jabber gateway Goal: - kick up simple-xmpp interoperability Agenda guidelines: - identify common issues of simple-xmpp interoperability - define best-practice solutions and workarounds of delicate issues - coding sessions in existing applications such as Asterisk, Openser, Jabberd, Freeswitch, eJabberd, libraries, client applications, etc. - testing sessions - reports about past experiences and results of the workshop Target participants: - developers of simple-xmpp products - people interested in testing simple-xmpp products - people interested in building simple-xmpp communication environments Cost: - free registration (everybody pays for its traveling and accommodation) Registration: - via e-mail at simple-xmpp(a)asipto.com - please write the motivation to participate and add bullets into agenda if you like to approach new subjects. The organizers reserve the right to select a group of people that will contribute most as this is mainly a workshop to approach issues, test interoperability and design new solutions. - updates about the event are posted at: http://www.asipto.com/index.php/simple-xmpp-developer-workshop-2008/ Size: - up to 20 people As of Jul 15, 2008, participating companies are: - INRIA - http://www.inria.fr - SIP-XMPP interoperability in Asterisk (http://www.asterisk.org) - Edvina - http://www.edvina.net - SIP-XMPP interoperability in Asterisk (http://www.asterisk.org) - AG Projects - http://www.ag-projects.com - MSRP-XMPP Interoperability (http://www.msrprelay.org) - Asipto - http://www.asipto.com - SIP-XMPP interoperability in Openser (http://www.openser.org) General discussions about sip-xmpp can be conducted via mailing list: sip-xmpp [at] xmpp.org http://mail.jabber.org/mailman/listinfo/sip-xmpp -- Daniel-Constantin Mierla http://www.asipto.com

16 years, 4 months

[Serusers] Limit call duration

by inge

Hi all, I would like to limit the call duration for two hours. Ideally, limitation could be drive by Seconds or Euros. Indeed, for example 10€ for mobile destination (0.1€/min for example) can allow the user to call during 100min. I have thinking of using SIPSAK to generate a BYE but I need something between SER and SIPSAK to do the calculation. What could be a workarround to do that ? Regards, Adrien .L

16 years, 4 months

[Serusers] PRACK on forked response

by Stephen Paterson

Hi all, I'm using SER purely for testing purposes just now. Currently testing our handling of forked responses and hit upon the following situation: My outgoing INVITE is forked to two UASs (also our own SIP endpoint), one of which sends a reliable 180 and the other an unreliable 180. The UAC sends a PRACK on the dialog ID of the reliable response but SER forwards this response on to both UASs the result being the UAS that sent the unreliable 180 responds to the PRACK with a 481 (the PRACK contains a To-tag that part defines a dialog unknown to that UAS). I confess I can't find anything describing this scenario in any RFC but it strikes me as incorrect behaviour on SERs part. Does SER support PRACK? If not then no matter, I can live with it - it doesn't affect the outcome of any call. If it does, does anyone have any thoughts on this? Cheers Steve Registered Address Lakeside, Bramley Road, Mount Farm, Milton Keynes, MK1 1PT, UK Registration No: 1397386 (Wales) P Please consider the environment and don't print this e-mail unless you really need to

16 years, 4 months

[OpenSER-Users] Registration redirect

by David Villasmil

Hello all! Lets say I have domain sip.xyz.com directed to IP 1.2.3.4. But I want customers to register and call usgin ip 9.8.7.6 which is NOT in the dns list for sip.xyz.com. If i'm not mistaken, doing a simple redirect I should be able to get my customers registered on 9.8.7.6. But is there any other considerations I should make? thanks! d

16 years, 4 months

[OpenSER-Users] Adam Kusmirek : nie ma mnie w pracy.

by Adam Kusmirek

Nie będzie mnie w pracy od 2008-07-14 i nie wrócę przed 2008-07-28. Podczas nieobecności zastępuje mnie Tomek Mazan.

16 years, 4 months

[OpenSER-Users] For Openser To Background

by Ali Jawad

Hi All When I start Openser it starts in the putty console and the output is displayed there, my softphone can logon during that time. If I do close that putty session my softphone can not logon anymore although if I do netstat it states that Openser is listening. Howe can I send openser to the background? Or make it work in daemon mode ? My config has this info debug=9 log_stderror=yes log_facility=LOG_LOCAL0 #fork=yes fork=no children=4 Thanks

16 years, 4 months

[Serusers] Hello, request of the module for RFC 4474 Identity authentication

by Ge Zhang

Hello, everyone, I would like to install SER2.0 with the module of RFC 4474 Identity authentication. Could you please tell me which module implemented this if you know? Thanks a lot! Best regards, Ge

16 years, 4 months

[OpenSER-Users] Openser - which b2bua to use ?

by Amit Jain

I have configured Openser and integrated with radius server. I have done auth and accounting too. Now i need some help for b2bua to terminate the session after prepaid credit is over. which is the recommended b2bua that i can use for my configuration. Please let me know, how to proceed further. Regards, Amit

16 years, 4 months

[OpenSER-Users] Contacts in 3xx message on one line or multiple lines?

by Hiers, David

Hi, Openser seems to be happy with multiple contacts on one line in a 302 message. Contact:<sip:5551234567@255.16.250.13:5060;user=phone>;q=0.5,<sip:555123 4567@255.16.250.10:5060;user=phone>;q=0.25 If the leftmost contact is down, the rightmost contact is tried as expected. However, Openser does not like multiple lines of one contact per line: Contact: <sip:5551234567@255.16.250.16:5060;dtg=SIP-NS> Contact: <sip:5551234567@255.16.250.19:5060;dtg=SIP-NS> Openser uses the topmost contact and ignores the bottommost contact in the multiple line list. If the topmost contact is down, the bottommost is never tried. Is Openser supposed to be able to handle multiline contact lists? If so, what are the common configuration issues that we can expect to see? Thanks, David Hiers CCIE (R/S, V), CISSP ADP Dealer Services 2525 SW 1st Ave. Suite 300W Portland, OR 97201 o: 503-205-4467 f: 503-402-3277 This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.

16 years, 4 months

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sr-users July 2008