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[Serusers] Improper relaying of Invite requests
by Kaushik Kumar 29 Nov '06

29 Nov '06

Hi, I am working on a project which involves testing SER with some dummy clients. I have downloaded and installed SER from iptel.org. I am using the very basic Hello.cfg configuration file available online. I find that when SER gets a INVITE request from the caller, it does not deliver it properly to the client (callee). Some debugging revealed that it relays the request to port 5060 at the client rather than the one which was registered by the callee earlier. I tried to look at the source code of SER and found that the callee's contact information never seem to be looked up at any stage. Is this due to some missing information in my configuration file? Any feedback would be appreciated. Thank you -Kaushik PS: I have attached (inline) the configuration file ser.cfg ------------------------------------ debug=9 fork=yes log_stderror=yes listen=128.42.3.3 port=5060 children=4 check_via=no # added dns=no rev_dns=no fifo="/tmp/ser_fifo" loadmodule "/scratch/ser-0.9.7-pre1/modules/sl/sl.so" # Stateless replies loadmodule "/scratch/ser-0.9.7-pre1/modules/tm/tm.so" # Transaction management loadmodule "/scratch/ser-0.9.7-pre1/modules/rr/rr.so" # Routing and record-routing loadmodule "/scratch/ser-0.9.7-pre1/modules/maxfwd/maxfwd.so" # Max fowards check loadmodule "/scratch/ser-0.9.7-pre1/modules/usrloc/usrloc.so" # User location support loadmodule "/scratch/ser-0.9.7-pre1/modules/registrar/registrar.so" # Registrar (requires userloc and sl) loadmodule "/scratch/ser-0.9.7-pre1/modules/textops/textops.so" # Message textual operations modparam("usrloc", "db_mode", 0) # 0 forces registration data to be stored in memory modparam("rr", "enable_full_lr", 1) route { # ------------------------------------------------------------------------ # Sanity Check Section # ------------------------------------------------------------------------ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ------------------------------------------------------------------------ # Record Route Section # ------------------------------------------------------------------------ if (method!="REGISTER") { record_route(); }; # ------------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------------ if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; # ------------------------------------------------------------------------ # Call Type Processing Section # ------------------------------------------------------------------------ if (uri!=myself) { append_hf("P-hint: outbound\r\n"); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; append_hf("P-hint: userloc applied\r\n"); route(1); } route[1] { # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ if (!t_relay()) { log(2, "Failed t_relay"); sl_reply_error(); }; } route[2] { # ------------------------------------------------------------------------ # REGISTER Message Handler # ------------------------------------------------------------------------ if (!save("location")) { sl_reply_error(); }; }

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Re: [Users] regular expression
by Jayesh Nambiar 29 Nov '06

29 Nov '06

Hi, Thanks for your reply but it will not match only two digits as I have specified the range( [0-9] ). Actually my primary requirement is it should not match any alphabets or special characters with the numbers. I want that only the number should be matched. w/regards, Jayesh ----- Original Message ---- From: Alexandr Dubovikov <shurik(a)start4.info> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: users(a)openser.org Sent: Wednesday, 29 November, 2006 1:15:51 PM Subject: Re: [Users] regular expression On Wed, Nov 29, 2006 at 12:58:44PM +0530, Jayesh Nambiar wrote: > Hi all, > Can someone please help me in formulating a regular expression. I have an expression like this: > if(uri=~"^sip:1[0-9]*@) > This matches numbers like 15552221212, but also matches number like 15552221212#. > > when i make it as: > if(uri=~"^sip:1[0-9]@*) not really, in this regexp your number can have only 2 digits and first one should ne "1". > Again this matches any special characters after the number. > > I just wanted a regular expression that will match only numbers and no alphabets or special characters. > Can someone please help me to formulate a regular expression for this. if(uri=~"^sip:1[0-9]+@*) > > Thanks in advance. -- Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info AD1-UANIC * ICQ: 122351182 * http://www.start4.info __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

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[Users] Complete Config
by Darryl Ross 29 Nov '06

29 Nov '06

Hey All, I am looking for someone who can give me some assistance with a building an openser configuration that is able to handle the following: 1) Multiple domains. 2) Mediaproxy for all clients (not just clients behind NAT). 3) RADIUS accounting for calls. 4) Least Cost Routing for selecting the cheapest upstream link. 5) Trusted (non-authenticated) upstream providers for receiving calls. 6) Call Redirection (all calls, busy, no answer) with redirected calls billed to the redirecting party. 7) Some network features, like voicemail, talking clock, etc, sent to an Asterisk machine. I currently have a configuration that does (1) through (5), but have had no luck in even getting call redirection to work, let alone getting the billing right. I'm having some issues, like: - Getting two replies to REGISTER messages which causes Asterisk clients to print a message "Got authentication request (401) on unknown REGISTER to <...>" - Sometimes I get a storm of CANCEL or BYE messages when a call is ending. I have absolutely no idea where the messages are coming from, whether its the client or upstream, and what message I'm sending that's causing it. The configuration I have made is just from reading the docs, looking at the howtos and tutorials and from guessing what should be done. I think the problems I'm having are related to not really understanding what should be happening 100% and having a config that is a lot more complicated than it needs to be. Anyone willing to give me some assistance? I can send a password sanitised copy of my current config directly if it would help. Regards Darryl

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[Users] strip
by Sam Hsu 29 Nov '06

29 Nov '06

i use the strip() function to strip the prefix when call out for example, prefix 0 to call out there are my sip invite message ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ INVITE sip:0928117466@202.111.222.76:5060 SIP/2.0 Record-Route: <sip:211.111.222.102;r2=on;lr=on;ftag=694064c4> Record-Route: <sip:211.111.222.102;transport=tcp;r2=on;lr=on;ftag=694064c4> Content-Length: 324 Content-Type: application/sdp Via: SIP/2.0/UDP 211.23.176.102;branch=z9hG4bK4eec.ecf680e4.0;i=1 Via: SIP/2.0/TCP 192.168.123.5:5060;received=220.132.138.7 ;branch=z9hG4bK69486617 To: <sip:00928117466@211.111.222.102> From: "joepass" <sip:joepass@211.111.222.102>;tag=694064c4 Supported: timer Call-ID: 95746504-39004973-1533c430-d8319d29(a)192.168.123.5 CSeq: 26589 INVITE User-Agent: IP SIP Phone/2.1.3 Max-Forwards: 69 Session-Expires: 1800 Allow: UPDATE,INFO,MESSAGE,REFER,NOTIFY,INVITE,ACK,OPTIONS,BYE,CANCEL Authorization: Digest nonce="456ba868fbd62c72ca16fcdd04678168a8fa0683", username="joepass", realm="votel-tech.com", uri=" sip:00928117466@211.111.222.102", response="73a8c869c2a42a12f0d920c2a7d6f068" P-IPRAuth: votel-tech.com Contact: <sip:joepass@220.111.222.7:1070> ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ in the example, the real number is 0928117466 add prefix 0, so the final number is 00928117466 I see the INVITE part the id has strip the prefix 0 but To: <sip:00928117466@211.111.222.102> and the Authorization part(uri) still keep on 00928117466 My gateway seems to use this information to call out. So it cause some error respond. If i call to the gateway directly(not through openser with number 0928117466), it works. How can i strip the "To" and "Authorization" part uri. thanks... Sam

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RE: [Users] OpenSER in front of Asterisk
by Thomas Deillon 29 Nov '06

29 Nov '06

Hi, We can use the dispatcher module, but it will only load balance the SIP traffic. Correct? I tried to use this http://mediaproxy.ag-projects.com/ with one ser server in front of two Asterisks. But I guess that this tool will only work with multiple ser servers... Is there a solution to make RTP/SIP load balancing with asterisks ? Thanks for your help, Thomas -----Message d'origine----- De : users-bounces(a)openser.org [mailto:users-bounces@openser.org] De la part de Daniel-Constantin Mierla Envoyé : vendredi, 13. octobre 2006 11:25 À : yusuf Cc : users(a)openser.org Objet : Re: [Users] OpenSER in front of Asterisk Hello, if you want to do load balncing, have a look at dispatcher module, it is intended to do that. Does not mather if behind are asterisks or other boxes. Cheers, Daniel On 10/10/06 10:49, yusuf wrote: > Hi all, > > I just joined this mailing list today :) > I'm sure you guys have heard this question before, but what I want to > do is put OpenSER in front of Asterisk to achieve load balancing. I > realise this entails more than it means, and will probably open up a > world of pain, I just want to know what others experiences of this > are, and any comments. > Also, with 1.1.0 being released, and with this new module called > 'path', will this be of any help > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Serusers] how to restrict one sipid login once at the same time
by liu york 29 Nov '06

29 Nov '06

Hello My means is that one sipid cant login again untel the same sipid logout. How to config that? Many thx. Warm regard!! 　　　　　　　　york 　　　　　　　　york(a)goldentek.biz

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[Users] OpenSER Summit 2006: summaries
by Daniel-Constantin Mierla 28 Nov '06

28 Nov '06

Hello, a bit late because of time constraints, and not yet fully complete -- I tried to punctuate the idea behind the talks and presentations at the summit, along with some personal conclusions. It should play along with the slides for those which were not able to attend. We have some short videos, but need to rework the audio. http://openser.blogspot.com/ The posting order is the same as events happened. The afternoon session of the second day will be reviewed soon. Best regards, Daniel

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[Serusers] SER Configuration with Mediatrix/Quintum VOIP Gateways
by Goran Donev 28 Nov '06

28 Nov '06

Can someone please give me an example configuration of how to set up ser to pass calls to pstn gateways? I want to set up SER to be able to pass calls based on certain dial plans to certain gateways I have set up. Please send me links or sample configuration files. It would be greatly appreciated. Thanks

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Re: [Serusers] Re: [Users] TM : retransmission timers
by Jiri Kuthan 28 Nov '06

28 Nov '06

Hi Samuel, thank you for your effort and great beginning. I see too this as a laborous effort but in case you decide to go on, you are most welcome to use the Ottendorf announcement as input. It is not complete, but the major changes are covered (hopefuly :-)) -jiri At 11:34 28/11/2006, samuel wrote: >Hi folks, >I've tried to stay out of this thread but I think it has gone far >beyond technical arguments and I don't think it will help neither SER >or openSER projects. I would just ask to stop this thread before it >get worst or stack to technical discussions (which I'm learning a lot) >and leave personal appreciations aside. > >I still see advantages on each project and depending on the >requirements, one is more suited than the other. I had the hope that >both would merge with the best of them but I see this almost >impossible right now... > >I'll try to summarize differences and I would apreciate *technical* >corrections to it (it's almost impossible to have an update >list....fortunately, i've tried to enumerate diferences in the dev >versions which will soon become stable): >*openSER: >o1)detailled documentation, including howtos >o2)fast reaction in the mailing list from the core developers >o3)incorporate user comments into new features >o4)decentralised developers >o5)improved usrloc (loading on mem, cacheless) >o6)gaining user acceptance->biggest testing comunity > >*SER >s1)improved TCP code >s2)improved timers >s3)easier integration with SEMS/SERWEB > >I have tried to be as opbjective as possible (but everyone knows it's >almost impossible) and it would be great to create a common place to >share experiences with both projects as someone has proposed but I >guess everyone is really really busy right now... > >Samuel. >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] B2BUA & Radius
by rupesh boyana 28 Nov '06

28 Nov '06

hi , i have integrated SER with NAT Helper +Media proxy + Mysql +PSTN modules. but i am unable to include B2BUA and Radius modules. can some one please guide me in this issue. moreover when i am calling a PSTN number from my SIP UA, after the call is connected, when the call is disconnected on the PSTN end, the BYE packet is not reaching my SIP UA end.similarly, when the call is cancelled at the PSTN end, it shows still RINGING in the SIP UA end. but when i call from SIP UA-1 to SIP UA-2 (online call) , when the call is terminated on either side, call gets disconnected on both sides. so, can some one throw some light on this ? moreover how r B2B UA and Radius related , i mean how do they communicate with each other? Thank you, Rupesh.

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