- sr-users - kamailio.org

[Serusers] Timer between Trying and 18x
by Fernando Schmitt 15 Aug '07

15 Aug '07

Guys, I modified some code in order do create a timer to control the time between Trying and 18x. Do you think it's usefull? In case you think, I would like to contribute. Fernando Schmitt

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[Serusers] SER Proxy with Redirect
by Fernando Schmitt 15 Aug '07

15 Aug '07

Juha, I'm new into C programming. Could you kindly let me know where this code should be inserted? yes, i implemented a function called move that turned 302 to invite, but that code relied on a patch to tm module that gave me access to contact header of reply in reply route. i found the function and it is below. someone who needs this functionality has to check if same kind of access to the contact header of the reply is still possible or if the access method is now different and modify the code accordingly. also, all contacts would need to be taken from 302, not just the first. contact_re was compiled during module initialization: regcomp(&contact_re, "^Contact:(.*)$", REG_EXTENDED|REG_NEWLINE); needless to say, i'm not proud of this code, but it gives an idea on how it can be done. -- juha /* * Appends a new branch to request using the first Contact URI of the (302) * reply (if any) as the Request URI. Returns 1 if appending succeeded and -1 * if it failed. */ int move(struct sip_msg* _m, char* _s1, char* _s2) { struct sip_msg* reply; regmatch_t pmatch[2]; struct hdr_field hf; contact_t* first; reply = _m->final_reply; if (!reply) { LOG(L_ERR, "move(): No reply found\n"); return -1; } LOG(L_ERR, "move(): unparsed part of reply: %s\n", reply->unparsed); if ((regexec(&contact_re, reply->unparsed, 2, &(pmatch[0]), 0) != 0) || (pmatch[1].rm_so == -1)) { LOG(L_ERR, "move(): No Contact header found\n"); return -1; } hf.type = HDR_CONTACT; hf.name.len = 0; hf.body.len = pmatch[1].rm_eo - pmatch[1].rm_so - 1; hf.body.s = &(reply->unparsed[pmatch[1].rm_so]); hf.len = hf.body.len + 2; hf.parsed = NULL; hf.next = NULL; LOG(L_ERR, "hf.body: '%.*s'\n", hf.body.len, hf.body.s); if (parse_contact(&hf) < 0) { LOG(L_ERR, "move(): Error while parsing Contact\n"); goto failure; } if (((contact_body_t*)hf.parsed)->star == 1) { LOG(L_ERR, "move(): Contact is *\n"); goto failure; } first = ((contact_body_t*)hf.parsed)->contacts; if (first) { if (append_branch(_m, first->uri.s, first->uri.len) == 1) { goto success; } else { LOG(L_ERR, "move(): Appending branch failed\n"); } } else { LOG(L_ERR, "move(): No contacts in Contact header\n"); } failure: if (hf.parsed) { free_contact((contact_body_t**)(&(hf.parsed))); } return -1; success: free_contact((contact_body_t**)(&(hf.parsed))); return 1; } Fernando Schmitt

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[OpenSER-Users] tm.so module
by Brian D 15 Aug '07

15 Aug '07

I am unable to call the t_relay_to_tcp() or t_relay_to_udp() functions even though the tm.so module is being loaded. Any thoughts? Thanks, - b

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[Serusers] Out of Office
by chayn＠imhocorp.com 15 Aug '07

15 Aug '07

Hello, I am out of the office and will be unable to respond to messages until Thursday, August 16. Claude

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[OpenSER-Users] Not insert Via header on forwarding/replicate/t_relay etc?
by Pat wang 15 Aug '07

15 Aug '07

Hi all, Is there anyway that I can use forward, t_replicate, or t_relay command without inserting a Via header? What I am trying to do is to use OpenSER as a load balancer but just filtering incoming messages and distribute to real server but not adding an extra Via header. Thanks for your help, Patrick _________________________________________________________________ Messenger Café open for fun 24/7. Hot games, cool activities served daily. Visit now. http://cafemessenger.com?ocid=TXT_TAGHM_AugHMtagline

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[OpenSER-Users] To configure TLS support
by Vikas CG. 15 Aug '07

15 Aug '07

Hi all, Can any help me out in running openser server with TLS support? In openser conf file they have commented for TLS and given comment that we must uncomment it to run TLS support but unfortunately while I compile it's giving configuration error, have I want to do some more settings. Thanks, Vikas C.G. -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Marc LEURENT Sent: Thursday, August 09, 2007 7:19 PM To: flavio Cc: users(a)openser.org Subject: Re: [OpenSER-Users] Re: How can it be simpler and working? CDRTool YEAHHHHHH! My problem was with the destination database. I didn't understand what was the destination ID, I thought it was just a random one! I was probably too tired... Hopefully, tomorrow; it's summer holidays I owe you a beer!! Are you going to Rome next September for VoIP conference? ShowPrice From=sip:101@sd-7501.dedibox.fr To=sip:001305465768 Duration=599876 199.9587 App: audio Destination: 1305 (TEST INTERNATIONAL) Customer: default Connect fee: 0.0000 -- Span: 1 Duration: 11931 s StartTime: 2007-08-09 15:41:09 (Europe/Paris) ProfileId: 441 for weekday RateId: 441 for 8-19h Rate: 0.0200 / 60 s Price: 3.9770 -- Span: 2 Duration: 29931 s StartTime: 2007-08-09 15:41:09 (Europe/Paris) ProfileId: 441 for weekday RateId: 442 for 19-24h Rate: 0.0200 / 60 s Price: 9.9770 -- Span: 3 Duration: 58731 s StartTime: 2007-08-09 15:41:09 (Europe/Paris) ProfileId: 441 for weekday RateId: 442 for 0-8h Rate: 0.0200 / 60 s Price: 19.5770 Yes: "reportMissingRates"=> 1 flavio a écrit : > Marc, > > Is the missing rates parameter of $CDRTool['rating'] equal to 1? > > rates.csv > Ops,Gateway,Domain,Subscriber,Rate Id,Destination,Price,App,Connect > 2,,192.168.1.160,,442,1305,200,audio, > 2,,192.168.1.160,,441,1305,200,audio, Ops,Gateway,Domain,Subscriber,Rate Id,Destination,Price,App,Connect 2,,sd-7501.dedibox.fr,,442,1305,200,audio, 2,,sd-7501.dedibox.fr,,441,1305,200,audio, > > > profiles.csv > Ops,Gateway,Domain,Subscriber,Profile Id,Rate Id1,00-H1,Rate > Id2,H1-H2,Rate Id3,H2-H3,Rate Id4,H3-24 > 2,,192.168.1.160,,441,442,8,441,19,442,24,,0 > 2,,192.168.1.160,,442,442,24,442,19,441,24,,0 Ops,Gateway,Domain,Subscriber,Profile Id,Rate Id1,00-H1,Rate Id2,H1-H2,Rate Id3,H2-H3,Rate Id4,H3-24 2,,sd-7501.dedibox.fr,,442,442,24,442,19,442,24,,0 2,,sd-7501.dedibox.fr,,441,442,8,441,19,442,24,,0 > > > customer.csv > > Ops,Gateway,Domain,Subscriber,WeekDay,Fallback,WeekEnd,Fallback,Timezone,Incr,Minim > > 2,,,,441,,442,,Europe/Amsterdam,0,0 Ops,Gateway,Domain,Subscriber,WeekDay,Fallback,WeekEnd,Fallback,Timezone,Incr,Minim 2,,,,441,,442,,Europe/Paris,0,0 > > > I have made all the tests dialing 001305...... (Miami). (I have used > 192.168.1.160 as the domain, change to yourdomain). It normalizes to > 1305 striping two zeros as specified on cdrlib.phtml. Check if it is > striping the digits to match the desired destination. > > class CDRS { > > var $CDR_class = 'CDR'; > var $intAccessCode = '00'; > var $natAccessCode = '0'; > class CDRS { var $CDR_class = 'CDR'; var $intAccessCode = '00'; var $natAccessCode = '0'; var $RatingEngineSocket = false; var $maxrowsperpage = 15; var $status = array(); var $normalizedField = 'Normalized'; var $DestinationIdField = 'DestinationId'; var $BillingIdField = 'UserName'; var $sipTraceDataSource = 'sip_trace'; var $defaults = array(); var $whereUnnormalized = ''; ...................... ...................... > Cheers, > > Flavio > ----- Original Message ----- From: "Marc LEURENT" <lftsy(a)free.fr> > To: "flavio" <flavio(a)asteriskguide.com> > Sent: Thursday, August 09, 2007 9:41 AM > Subject: Re: [OpenSER-Users] Re: How can it be simpler and working? CDRTool > > >> Hello flavio, >> I would be very grateful if you can send me just the .csv of your >> tables about your internal calls, the simpliest one for you that is >> working... >> Best Regards! >> >> Marc, desperate >> >> Marc LEURENT a écrit : >>> I've got a running tail -f /var/lig/messages and I haven't seen a >>> single error... >>> It's just that it calculates 0.0000 for the price >>> >>> I have already added the $CDRTool['rating'] parameter.. >>> >>> Id don't have the same message: >>> Aug 9 14:00:02 sd-7501 CDRTool[18946]: ShowPrice Id=3 >>> Timestamp=1186660750 Duration=11 inputTraffic=0 outputTraffic=0 >>> From=sip:106@sd-7501.dedibox.fr >>> To=6713(a)sd-7501.dedibox.fr Gateway= Application=audio >>> Aug 9 14:00:02 sd-7501 CDRTool[19627]: Normalization done in 0 s, >>> memory usage: 9.90 MB >>> Aug 9 14:00:02 sd-7501 CDRTool[19627]: 1 CDRs, 1 normalized in 0 s @ >>> 0 cps >>> >>> But there is no "ConnectFee=0.0000 Span=1 Duration=57 De >>> stId=1305 domain=192.168.1.160 Profile=441 Period=weekday Rate=441 >>> Interval=8 >>> -19 Cost=0.0200/10 Price=0.1140 " after >>> >>> Do I have to activate something else?? >>> >>> >>> flavio a écrit : >>>> Marc, >>>> >>>> I'm using 5.1.5 and it is working. Check the /var/log/syslog for errors >>>> in the normalization process >>>> >>>> Something like: >>>> >>>> Aug 6 07:18:55 debian CDRTool[3973]: Error: Cannot find rates for >>>> callid=, d >>>> omain=192.168.1.160, destination=1305, profile=441, app=audio >>>> >>>> The correct in the tests I have made is: >>>> >>>> Aug 6 07:47:44 debian CDRTool[3973]: ShowPrice Id=88 >>>> Timestamp=1186400781 Du >>>> ration=57 inputTraffic=0 outputTraffic=0 From=1000(a)192.168.1.160 >>>> To=001305345 >>>> 6789(a)192.168.1.160 Gateway=192.168.1.118 Application=audio >>>> Aug 6 07:47:44 debian CDRTool[3973]: ConnectFee=0.0000 Span=1 >>>> Duration=57 De >>>> stId=1305 domain=192.168.1.160 Profile=441 Period=weekday Rate=441 >>>> Interval=8 >>>> -19 Cost=0.0200/10 Price=0.1140 >>>> >>>> please activate the following parameter in the global.inc file >>>> (reportMissingRates). >>>> >>>> $CDRTool['rating']=array("priceDenominator" => >>>> 10000,"priceDecimalDigits" => 4,"minimumDurationCharged"=> >>>> 0,"durationPeriodRated"=> 60,"trafficSizeRated"=> >>>> 1024,"reportMissingRates"=> 1,"minimumDuration"=> 0); >>>> >>>> I hope it helps, it took me a long time before to start working. >>>> >>>> Cheers >>>> >>>> Flavio >>>> >>>> >>>> ----- Original Message ----- From: "Marc LEURENT" <lftsy(a)free.fr> >>>> To: "flavio" <flavio(a)asteriskguide.com> >>>> Cc: <users(a)openser.org> >>>> Sent: Thursday, August 09, 2007 9:01 AM >>>> Subject: Re: [OpenSER-Users] Re: How can it be simpler and working? >>>> CDRTool >>>> >>>> >>>>> Yes, I've added all rights to my account... >>>>> and price is NULL in the database until a Normalize is done! All >>>>> prices are set to 0.0000 after... >>>>> >>>>> And after a >>>>> Normalize datasource ser_radius, database DB_radius, table >>>>> radacct200708 >>>>> Aug 9 14:00:02 sd-7501 CDRTool[18946]: ShowPrice Id=3 >>>>> Timestamp=1186660750 Duration=11 inputTraffic=0 outputTraffic=0 >>>>> From=sip:106@sd-7501.dedibox.fr >>>>> To=6713(a)sd-7501.dedibox.fr Gateway= Application=audio >>>>> >>>>> >>>>> >>>>> >>>>> flavio a écrit : >>>>>> Marc, >>>>>> >>>>>> Have you checked your user rights in the CDRTool (is show price item >>>>>> highlighted?). My CDRtool shown x.xxx in the price and that's was >>>>>> the >>>>>> cause. >>>>>> >>>>>> Flavio >>>>>> >>>>>> ----- Original Message ----- From: "Marc LEURENT" <lftsy(a)free.fr> >>>>>> To: "Dan-Cristian Bogos" <dan.bogos(a)gmail.com> >>>>>> Cc: <users(a)openser.org> >>>>>> Sent: Thursday, August 09, 2007 8:01 AM >>>>>> Subject: [OpenSER-Users] Re: How can it be simpler and working? >>>>>> CDRTool >>>>>> >>>>>> >>>>>> I'm starting to fell sorry to bother you like that... >>>>>> I have changed the database like you said, but it is still not >>>>>> working... >>>>>> >>>>>> ShowPrice From=sip:101@sd-7501.dedibox.fr To=0033677832974 >>>>>> Duration=5923 >>>>>> 0.0000 >>>>>> >>>>>> Should I try to downgrade to version 5.1.3 like yours? >>>>>> >>>>>> ShowMemory >>>>>> 1 destinations >>>>>> 1 customers >>>>>> 1 profiles >>>>>> 1 rates >>>>>> 0 holidays >>>>>> 2 prepaid accounts >>>>>> 1 sessions >>>>>> 1 sessions last minute >>>>>> 1 sessions last hour >>>>>> 1 sessions last day >>>>>> >>>>>> >>>>>> but in the database, the prepaid account is credited by 10.00 >>>>>> ShowPrepaidAccounts Filter=101(a)sd-7501.dedibox.fr >>>>>> Account=101(a)sd-7501.dedibox.fr Balance=10040.0000 >>>>>> >>>>>> ShowCustomers >>>>>> default >>>>>> >>>>>> >>>>>> ShowAccounts >>>>>> Error: Invalid request >>>>>> >>>>>> MaxSessionTime From=sip:101@sd-7501.dedibox.fr >>>>>> To=sip:106@sd-7501.dedibox.fr Duration=7200 Lock=1 >>>>>> none >>>>>> >>>>>> DebitBalance From=sip:101@sd-7501.dedibox.fr >>>>>> To=sip:106@sd-7501.dedibox.fr Duration=59 >>>>>> Ok >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Dan-Cristian Bogos a écrit : >>>>>>> Ok, let's take it in your way. I would rather modify your data into: >>>>>>> >>>>>>> Destinations: >>>>>>> Ops,Gateway,Domain,Subscriber,Destination Id,Description >>>>>>> 2,,,,131,TEST INTERNATIONAL >>>>>>> >>>>>>> Customers: >>>>>>> Ops,Gateway,Domain,Subscriber,WeekDay,Fallback,WeekEnd,Fallback,Timezone,Incr,Minim >>>>>>> >>>>>>> >>>>>>> >>>>>>> 2,,,,441,,441,,,1,1 >>>>>>> >>>>>>> Profiles: >>>>>>> Ops,Gateway,Domain,Subscriber,Profile Id,Rate Id1,00-H1,Rate >>>>>>> Id2,H1-H2,Rate Id3,H2-H3,Rate Id4,H3-24 >>>>>>> 2,,,,441,441,8,441,19,441,24,441,0 >>>>>>> >>>>>>> Rates: >>>>>>> Ops,Gateway,Domain,Subscriber,Rate Id,Destination,Price,App,Connect >>>>>>> 2,,,,441,131,2100,audio,0 >>>>>>> >>>>>>> U did not have a rate id and no App. Without App field for me it >>>>>>> does >>>>>>> not calculate either. >>>>>>> >>>>>>> DanB >>>>>>> >>>>>>>> >>>>>>>> But, the command ShowPrice returns: >>>>>>>> ShowPrice From=sip:101@sd-7501.dedibox.fr To=0033677832974 >>>>>>>> Duration=59 >>>>>>>> 0.0000 >>>>>>>> >>>>>>>> >>>>>>>> I have a field TrafficRate in the database but doesn't appear on >>>>>>>> the >>>>>>>> CDRTool interface >>>>>>>> I put random values inside >>>>>>>> >>>>>>>> >>>>>>>> Dan-Cristian Bogos a écrit : >>>>>>>>> My version is: >>>>>>>>> CDRTool version 5.1.3 >>>>>>>>> >>>>>>>>> In my rates table I have one more field more which is >>>>>>>>> TrafficRate. I >>>>>>>>> see that this one is missing on your side. >>>>>>>>> Can u also create a default rate table without filling any of the >>>>>>>>> gateway, subscriber, domain in profiles? This one should match all >>>>>>>>> the >>>>>>>>> destinations. Maybe would be useful till u make it work to have >>>>>>>>> only >>>>>>>>> the test row and remove anything else to take it gradually. >>>>>>>>> Also, try querying for price with 00 added in the front of the >>>>>>>>> destination since this is the default prefix set in CDRTool for >>>>>>>>> international destinations (without being necessary to be added in >>>>>>>>> rate or destination table). >>>>>>>>> >>>>>>>>> DanB >>>>>>>>> >>>>>>>>> On 8/9/07, Marc LEURENT <lftsy(a)free.fr> wrote: >>>>>>>>>> I have added: >>>>>>>>>> >>>>>>>>>> Destination: >>>>>>>>>> Ops,Gateway,Domain,Subscriber,Destination Id,Description >>>>>>>>>> 2,,sd-7501.dedibox.fr,,31800,Ma Dst TEST >>>>>>>>>> 2,,,33677832974,131,TEST INTERNATIONAL >>>>>>>>>> >>>>>>>>>> Profiles: >>>>>>>>>> Ops,Gateway,Domain,Subscriber,Profile Id,Rate Id1,00-H1,Rate >>>>>>>>>> Id2,H1-H2,Rate Id3,H2-H3,Rate Id4,H3-24 >>>>>>>>>> 2,,sd-7501.dedibox.fr,101(a)sd-7501.dedibox.fr,441,441,8,441,19,441,24,441,0 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> 2,,sd-7501.dedibox.fr,106(a)sd-7501.dedibox.fr,441,441,8,441,19,441,24,441,0 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> 2,,,33677832974,441,441,22,441,23,441,24,441,0 >>>>>>>>>> >>>>>>>>>> Rates: >>>>>>>>>> Ops,Gateway,Domain,Subscriber,Rate >>>>>>>>>> Id,Destination,Price,App,Connect >>>>>>>>>> 2,,sd-7501.dedibox.fr,,441,31800,2100,audio,4647 >>>>>>>>>> 2,,sd-7501.dedibox.fr,101(a)sd-7501.dedibox.fr,441,31800,200,audio,2364 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> 2,,sd-7501.dedibox.fr,102(a)sd-7501.dedibox.fr,441,31800,200,audio,245 >>>>>>>>>> >>>>>>>>>> 2,,,33677832974,441,131,2100,,2426 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> ShowPrice From=sip:101@sd-7501.dedibox.fr To=33677832974 >>>>>>>>>> Duration=59 >>>>>>>>>> 0.0000 >>>>>>>>>> >>>>>>>>>> But it is still not working... >>>>>>>>>> Could you spent some time sending me a part of yours... >>>>>>>>>> And are you using CDRTool 5.2.1?? >>>>>>>>>> >>>>>>>>>> Thanks >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Dan-Cristian Bogos a écrit : >>>>>>>>>>> Hi Marc, >>>>>>>>>>> >>>>>>>>>>> can u do me a favor and try adding in an international >>>>>>>>>>> destination in >>>>>>>>>>> your pricelist and try placing a call to it. See if you get >>>>>>>>>>> billing >>>>>>>>>>> calculation for it. I suspect that your billing doesn't match >>>>>>>>>>> the >>>>>>>>>>> destination due to prefixes which are normally stripped. Not >>>>>>>>>>> quite a >>>>>>>>>>> CDRTool internals expert but my installation simply works for >>>>>>>>>>> international destinations. >>>>>>>>>>> >>>>>>>>>>> DanB >>>>>>>>>>> >>>>>>>>>>> On 8/9/07, Marc LEURENT <lftsy(a)free.fr> wrote: >>>>>>>>>>>> Hello Dan-Cristian, >>>>>>>>>>>> >>>>>>>>>>>> I just want a simple thing, that when I'm connecting to the >>>>>>>>>>>> rating engine using telnet: telnet 127.0.0.1 9024 >>>>>>>>>>>> When I enter such a command, that the price should be != 0: >>>>>>>>>>>> >>>>>>>>>>>> ShowPrice From=sip:101@sd-7501.dedibox.fr >>>>>>>>>>>> To=sip:106@sd-7501.dedibox.fr Duration=59 >>>>>>>>>>>> 0.0000 >>>>>>>>>>>> >>>>>>>>>>>> Help would be very grateful... >>>>>>>>>>>> Thanks... >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> The rating engine is requesting the good radaact200708 table (I >>>>>>>>>>>> have change my globac.inc conf file >>>>>>>>>>>> "ser_radius"=>array( >>>>>>>>>>>> "name" => "OpenSER", >>>>>>>>>>>> "class" => "CDRS_ser_radius", >>>>>>>>>>>> "table" => >>>>>>>>>>>> "radacct".date("Ym"), >>>>>>>>>>>> ...... >>>>>>>>>>>> >>>>>>>>>>>> ) >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Here is a view of the rating tables... And I don't understand >>>>>>>>>>>> where I'm wrong.. >>>>>>>>>>>> >>>>>>>>>>>> Destinations: >>>>>>>>>>>> Ops,Gateway,Domain,Subscriber,Destination Id,Description >>>>>>>>>>>> 2,,sd-7501.dedibox.fr,,31800,Ma Dst TEST >>>>>>>>>>>> >>>>>>>>>>>> Customers: >>>>>>>>>>>> Ops,Gateway,Domain,Subscriber,WeekDay,Fallback,WeekEnd,Fallback,Timezone,Incr,Minim >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> 2,,sd-7501.dedibox.fr,101(a)sd-7501.dedibox.fr,441,,442,,Europe/Paris,20,1 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> 2,,sd-7501.dedibox.fr,106(a)sd-7501.dedibox.fr,441,,442,,Europe/Paris,34,1 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Profiles: >>>>>>>>>>>> Ops,Gateway,Domain,Subscriber,Profile Id,Rate Id1,00-H1,Rate >>>>>>>>>>>> Id2,H1-H2,Rate Id3,H2-H3,Rate Id4,H3-24 >>>>>>>>>>>> 2,,sd-7501.dedibox.fr,,441,442,8,441,19,441,24,441,0 >>>>>>>>>>>> 2,,sd-7501.dedibox.fr,,442,442,24,442,0,442,0,442,0 >>>>>>>>>>>> >>>>>>>>>>>> Rates: >>>>>>>>>>>> Ops,Gateway,Domain,Subscriber,Rate >>>>>>>>>>>> Id,Destination,Price,App,Connect >>>>>>>>>>>> 2,,,,441,31800,2100,audio,0 >>>>>>>>>>>> 2,,,,442,31800,200,audio,0 >>>>>>>>>>>> >>>>>>>>>>>> Prepaid: >>>>>>>>>>>> Ops,Account,Balance,Last change,Lock,Start time,Last >>>>>>>>>>>> price,Max,Destination >>>>>>>>>>>> 2,101(a)sd-7501.dedibox.fr,10030.0000,2007-08-08 >>>>>>>>>>>> 17:57:00,,0000-00-00 00:00:00,0.0000,0, >>>>>>>>>>>> 2,106(a)sd-7501.dedibox.fr,20010.0000,2007-08-08 >>>>>>>>>>>> 17:38:09,,0000-00-00 00:00:00,0.0000,0, >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> ShowPrepaidAccounts Filter=sd-7501.dedibox.fr >>>>>>>>>>>> Account=101(a)sd-7501.dedibox.fr Balance=10030.0000 >>>>>>>>>>>> Account=106(a)sd-7501.dedibox.fr Balance=20010.0000 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> And here is my radacct200708 table: >>>>>>>>>>>> RadAcctId AcctSessionId AcctUniqueId UserName >>>>>>>>>>>> Realm NASIPAddress NASPortId NASPortType >>>>>>>>>>>> AcctStartTime AcctStopTime AcctSessionTime >>>>>>>>>>>> AcctAuthentic ConnectInfo_start ConnectInfo_stop >>>>>>>>>>>> AcctInputOctets AcctOutputOctets CalledStationId >>>>>>>>>>>> CallingStationId AcctTerminateCause >>>>>>>>>>>> ServiceType FramedProtocol FramedIPAddress AcctStartDelay >>>>>>>>>>>> AcctStopDelay SipMethod SipResponseCode SipToTag >>>>>>>>>>>> SipFromTag >>>>>>>>>>>> SipTranslatedRequestURI SipUserAgents >>>>>>>>>>>> SipApplicationType SipCodecs SipRPID >>>>>>>>>>>> SipRPIDHeader SourceIP SourcePort CanonicalURI DelayTime >>>>>>>>>>>> Timestamp DestinationId Rate Price Normalized >>>>>>>>>>>> BillingId MediaInfo RTPStatistics FromHeader >>>>>>>>>>>> UserAgent Contact >>>>>>>>>>>> >>>>>>>>>>>> "2";"vjyxdmwkbbjajnc(a)192.168.95.50";"336224af44cd989a";"sip:106@sd-7501.dedibox.fr";"sd-7501.dedibox.fr";"88.191.45.91";"5060";;"2007-08-09 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> 10:55:58";"2007-08-09 >>>>>>>>>>>> 10:56:03";"5";NULL;NULL;;"0";"0";;;"200";"Sip-Session";;;"0";"0";"INVITE";"200";"c0a80101-20e94fd";"nxitc";;;;;;;;;"101(a)sd-7501.dedibox.fr";;"0";;;NULL;"0";;NULL;;;; >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> And my syslog: >>>>>>>>>>>> >>>>>>>>>>>> Aug 9 11:00:03 sd-7501 CDRTool[17377]: Normalize datasource >>>>>>>>>>>> ser_radius, database DB_radius, table radacct200708 >>>>>>>>>>>> Aug 9 11:00:03 sd-7501 CDRTool[17377]: Lock >>>>>>>>>>>> ser_radius:radacct200708 >>>>>>>>>>>> Aug 9 11:00:03 sd-7501 CDRTool[17097]: ShowPrice Id=2 >>>>>>>>>>>> Timestamp=1186649758 Duration=5 inputTraffic=0 outputTraffic=0 >>>>>>>>>>>> From=sip:106@sd-7501.dedibox.fr >>>>>>>>>>>> To=101(a)sd-7501.dedibox.fr Gateway= Application=audio >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users(a)openser.org >>>>>> http://openser.org/cgi-bin/mailman/listinfo/users >>> > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Serusers] sip express router for windows
by Youngjin Park 15 Aug '07

15 Aug '07

Is there a sip express router for windows? If not, do you have any plan in near future? Thanks. Young

2 1

[Serusers] record_route() & record_route_preset() di
by Kostas Marneris 15 Aug '07

15 Aug '07

Hello, I have a question regarding record_route() & record_route_preset() functions. I faced up a problem with some SIP clients with inbound calls where I believe that it's related with these functions. 1/ When I use the "record_route()" function in a single server SER setup I have the following "Record-Route:" header in INVITE mesg from SER to called SIP UA: Record-Route: <sip:1.2.3.4;ftag=AAA-19972;lr=on> ... 2/ When I use the "record_route_preset()" function in a double server SER setup (2 SER servers behind a Cisco CSM Load balancer), example : record_route_preset("5.6.7.8"); I have the following "Record-Route:" header in INVITE mesg (SER -> called SIP UA): Record-Route: <sip:1234567890@5.6.7.8;ftag=XPH-5307;lr=on> ... In first case the SIP UA accepts the inbound call and everything is Ok. In second case the same SIP UA doesn't replied with '100 Trying' mesg back to SER. In the second case, SER also adds the called number "1234567890" in the Record-Route header. Why there is this difference and if that can cause problems (incompatibility) to some SIP clients ? thanks for any hint, Kostas

2 1

[Serusers] ser asterisk pstn gateway
by ronald＠stanford.edu 15 Aug '07

15 Aug '07

hello all, I am stuck and need your help for +xlite +ser +asterisk +pstn calls. I dial 914082743304 and get "caller Not Found". I am able to authenticate and complete sip -> sip calls, but cannot complete PSTN calls I did monitor asterisk and no reroute was offered. please help! permissions.allow: ^sip:91[0-9]{10}@bigu\.edu$ ^sip:9011[0-9]*@bigu\.edu$ ^sip:9[0-9]{10}@bigu\.edu$ ngrep -vvvv proto udp port 5060 produces the following response: 171.66.176.123:47729 -> 192.168.11.243:5060 INVITE sip:914082743304@bigu.edu SIP/2.0..Via: SIP/2.0/UDP 192.168.1.100:47729;br anch=z9hG4bK-d87543-4f5eb56fc40f1f43-1--d87543-;rport..Max-Forwards: 70..Contact: <si p:cell@192.168.1.100:47729>..To: "914082743304"<sip:914082743304@bigu.edu>..From: "SER EXpress"<sip:cell@bigu.edu>;tag=b9283f27..Call-ID: Y2JkY2ZmMDQyMjQ3YTljYTg5 NTdlOGFiNGUyZWVlY2Q...CSeq: 1 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER , NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..User-Agent: eyeBea m release 1011s stamp 41121..Content-Length: 421....v=0..o=- 1 2 IN IP4 192.168.1.100 ..s=CounterPath eyeBeam 1.5..c=IN IP4 192.168.1.100..t=0 0..m=audio 32942 RTP/AVP 107 100 106 6 0 105 8 18 3 5 101..a=fmtp:18 annexb=yes..a=fmtp:101 0-15..a=rtpmap:107 BV 32/16000..a=rtpmap:100 SPEEX/16000..a=rtpmap:106 SPEEX-FEC/16000..a=rtpmap:105 SPEEX- FEC/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000..a=sendrecv..a=x-r tp-session-id:AEB31A44A04C4B32BD3D72349E4FBB80.. # U 192.168.11.243:5060 -> 171.66.176.123:47729 SIP/2.0 404 caller Not Found..Via: SIP/2.0/UDP 192.168.1.100:47729;branch=z9hG4bK-d875 43-4f5eb56fc40f1f43-1--d87543-;rport=47729;received=171.66.176.123..To: "914082743304 "<sip:914082743304@bigu.edu>;tag=21f9bfca23409407a263e4a7e40abe9d.f2a8..From: "SE R EXpress"<sip:cell@bigu.edu>;tag=b9283f27..Call-ID: Y2JkY2ZmMDQyMjQ3YTljYTg5NTdl OGFiNGUyZWVlY2Q...CSeq: 1 INVITE..Server: Sip EXpress router (0.9.4 (i386/linux))..Co ntent-Length: 0..Warning: 392 192.168.11.243:5060 "Noisy feedback tells: pid=15307 re q_src_ip=171.66.176.123 req_src_port=47729 in_uri=sip:914082743304@bigu.edu out_u ri=sip:914082743304@bigu.edu via_cnt==1".... debug=3 fork=yes log_stderror=no listen=1xx.x4.11.xx3 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 alias=sip2.bigu.edu dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") #added these timers for voicemail forwarding modparam("tm", "fr_inv_timer", 25) modparam("tm", "noisy_ctimer", 1) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") 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=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("1xx.x4.11.xx3:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("sip2.bigu.edu","subscriber")) { proxy_challenge("sip2.bigu.edu","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; break; }; if (method=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("sip2.bigu.edu","subscriber")) { www_challenge("sip2.bigu.edu","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!allow_trusted()) { if (!proxy_authorize("sip2.bigu.edu","subscriber")) { proxy_challenge("sip2.bigu.edu","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (uri=~"^sip:1[0-9]{10}@bigu.edu") { strip(1); }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); break; }; if (uri=~"^sip:011[0-9]*@bigu.edu") { # International PSTN route(4); route(5); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@bigu.edu") { # Domestic PSTN route(4); route(5); break; }; sl_send_reply("404", "caller Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("1xx.x4.8.xx8"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

1 0

[OpenSER-Users] openser 1.2.2 scheduled for Thursday, August 16, 2007
by Daniel-Constantin Mierla 14 Aug '07

14 Aug '07

Hello, we plan a minor release of 1.2.2 for Thursday, August 16, 2007. It was discussed on the last irc meeting. If you know a critical bug that should stop this release, please let us know. The ones registered on the tracker will be reviewed and considered for final decision of 1.2.2, don't send them again. Cheers, Daniel

2 1

[OpenSER-Users] Saving public IP in CDR
by Andreas Sikkema 14 Aug '07

14 Aug '07

Hi, I'm trying to do add accounting to our OpenSER server. Getting accounting to work is easy but I'm having trouble saving client IP addresses with every CDR. Most of our SIP UAs are behind some form of NAT and to "prove" that calls have originated at the client I want to save the public IP address and as extra service also the private IP address of the UA. Saving the private IP address is easy, $Ri contains this value. I'm having trouble finding out what to do to save the public IP address. I thought that since I'm using fix_contact() I could save the contact header body, but $ct still contained the private IP address. Is this even possible? -- Andreas Sikkema

2 2

[OpenSER-Users] New H350 Module
by Christian Schlatter 14 Aug '07

14 Aug '07

Hello everybody! A new openser module named H350 is available in the openser devel repository. http://www.openser.org/docs/modules/devel/h350.html This module allows a script writer to implement H.350 based SIP digest authentication, call forwarding, SIP URI alias to AOR rewriting, and service level parsing. H.350 is an ITU-T Recommendation that standardizes how real-time communications (RTC) account data is stored in an LDAP directory. In particular, H.350.4 defines LDAP object classes for storing SIP account data. H.350 and H.350.4 are also described in IETF RFC 3944. The H350 module uses the LDAP module API to query H.350 directory servers, and it exports script functions to parse H.350 LDAP attributes into openser AVPs. This allows to integrate openser proxies with an enterprise/provider's existing identity management infrastructure. The module documentation should provide all information needed to use this module with your openser installation. Feel free to ask questions regarding this module on this list. /Christian - University of North Carolina

1 0

[OpenSER-Users] Click 2 Call / click To Call applications
by liran tal 14 Aug '07

14 Aug '07

Hey everyone, Following a very interesting but old and un-answered post here: http://www.voipuser.org/forum_topic_6616.html about the integration of Click-To-Call web applications with OpenSER I wanted to give it a try myself through the FIFO interface. Before jumping into scripts and code I was wondering if someone can make it clear how to use those fifo commands that "initiate" a call between 2 parties, from that post, the user uses: /* construct FIFO command */ $fifo_cmd=":t_uac:".$myfilename."\n". "REFER\n".$caller."\n". "p-version: ".$signature."\n". "Contact: ".$web_contact."\n". "Referred-By: ".$web_contact."\n". "Refer-To: ".$callee."\n". "\n". /* EoHeader */ ".\n\n"; /* EoFifoRequest */ Is there documentation available on the various fifo commands like this that I can use? Can I simply echo the fifo command string to /tmp/openser_fifo? Thanks in advance, Regards, Liran.

2 5

[OpenSER-Users] Logging failed registration attempt
by Edoardo Serra 14 Aug '07

14 Aug '07

Hi all, I'd like to log failed SIP REGISTER attempt either with xlog or with sip_trace() but I cannot understand where to put related code to catch the authentication error Here is the part of my opensr.cfg dedicated to REGISTER handling if (method=="REGISTER") { if (!proxy_authorize("exorsa", "openser_view")) { proxy_challenge("exorsa", "0"); exit; } if (!check_to()) { sl_send_reply("403", "Digest username and URI username do NOT match! Stay away!"); exit; } save("location"); exit; }; Tnx in advance Regards Edoardo

3 6

[OpenSER-Users] OpenSER coredumps on acc.so module
by liran tal 14 Aug '07

14 Aug '07

Hey everyone, I've enabled RADIUS accounting in my config file and setup radiusclient.confand the servers file with the appropriate data though OpenSER coredumps for some reason, this is the backtrace from gdb: #0 0xb7afc168 in test_config (rh=0x824d450, filename=0x824b240 "/var/csis/cfg/radiusclient.conf") at config.c:389 389 config.c: No such file or directory. in config.c (gdb) backtrace #0 0xb7afc168 in test_config (rh=0x824d450, filename=0x824b240 "/var/csis/cfg/radiusclient.conf") at config.c:389 #1 0xb7afbf49 in rc_read_config (filename=0x824b240 "/var/csis/cfg/radiusclient.conf") at config.c:312 #2 0xb7b1440b in extra2strar () from /usr/local/lib/openser/modules/acc.so #3 0x0808dab1 in init_child () #4 0x0806d15b in main () servers file: 192.168.0.1 test123 the servers file was before with a key of some non-standard chars like "test123!!5%^#" so I changed it to "test123" thinking that this might have caused it but this is not the reason. Thanks in advnace, Lir.

2 2

[OpenSER-Users] Werid records in 'presentity'table when PUBLISHing
by KevinKinnan 14 Aug '07

14 Aug '07

If the Route Header(sip:[ip add]:[port]) or Outbound Proxy(sip:[ip add]:[port]) in the PUBLISH request has no Lr parameter, the username field in 'presentity' table is null, and the domain field is also wrong. Without Lr parameter(no loose routing), the request received by Presence server is that placing the remote target URI into the Request URI(rfc 3561), and adding a Route Header with the value of the remote target URI. It seems the presence module just takes the Request-URI(which actually is the address of presence server) in the received request and save username and domain into the database. Thus, the username is null and the domain is the address of presence server. Is that right?? Why doesn't this happen in 1.2.0 version?THANK YOURegards,Kevin _________________________________________________________________ 通过 Windows Live Spaces 与朋友轻松共享您的生活。 http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

2 1

[OpenSER-Users] a=nortpproxy:yes in SDP and fix_nated_*
by liran tal 14 Aug '07

14 Aug '07

Hey everyone, When I'm doing sequential forking I'm seeing that for each gateway that doesn't answer and I move forward in the list of available gateways to forward the INVITE to, openser adds an "a=nortpproxy:yes" into the SDP so what happens is that the final gateway which is alive receives the sip header with multiple lines of "a=nortpproxy:yes" repeating themselves in the SDP and throws away the packet. >From my browsing in nathelper's code I see the only place that writes a "a=nortpproxy:yes" string to the SDP is the fix_nated_sdp function, and I'm actually not using it in my cfg but rather using fix_nated_register and fix_nated_contact which from what the README says are not handling the SDP part. And so I'm asking, where could it be coming from? And also, it would be easier if I can insert a piece of code that can do some regular expression miracles with the SIP header and remove any duplicate lines of the "nortpproxy" string. I'm running on openser 1.1.1 Thanks, Lir.

2 2

[Serusers] Out of Office
by chayn＠imhocorp.com 14 Aug '07

14 Aug '07

Hello, I am out of the office and will be unable to respond to messages until Thursday, August 16. Claude

1 0

[OpenSER-Users] [avp_db_query] How to analize resulting AVP's row by row?
by Iñaki Baz Castillo 14 Aug '07

14 Aug '07

Hi, in order to implement a ACL system for incoming calls I've created a table "acl" with those columns: - priority - from_username - from_domain - to_username - group_from - group_to - type | action In that tables I store rules, and my idea is to work as a firewall: the first match is ejecuted (action=ALLOW or action=REJECT) and exit. But I can't do it since I don't know how to use all the AVP's JUST for the first row, then for the second: avp_db_query("SELECT priority,from_username,from_domain,to_username,group_from,group_to,type,action FROM acl", "$avp(s:priority);$avp(s:from_username);$avp(s:from_domain);$avp(s:to_username);$avp(s:group_from);$avp(s:group_to);$avp(s:type);$avp(s:action)"); This query results in many rows, but I can't analize each row, I just can compare each AVP with a value, but for that AVP they are possible all the values of any row. Because this I'm thinking in using the "exec" module to call a system script. Is not possible to access to tjhe query result by rows? Thanks for any help. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 4

[OpenSER-Users] Failure Route - 408 Timeout & Asterisk
by Kelvin Williams 13 Aug '07

13 Aug '07

I would like to advise my users as to the reason the phone stopped ringing (no, I'm not going to let it ring for 20 minutes as many of them wish that I would). My grand design is to send the call to Asterisk which plays a recording to the effect of, "the party you have dialed does not answer, please try your call later" after the 2 minute timeout. Upon 408 the call is sent to Asterisk, however there is no audio. The call stays alive until Asterisk hits the Hangup() in the dialplan. I've tested by adding extra audio (to make sure the call does stay up) and watching on the Asterisk console. In other situations, audio passes fine between the Asterisk and the endpoint (ie. Calling voicemail). Any advice would be greatly appreciated. Some relevant portions of the config: loadmodule "tm.so" modparam("tm", "fr_timer", 3) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 10) modparam("tm", "fr_inv_timer_avp", "$avp(s:timeout)") route[ROUTE_TO_PSTN] { .SNIP. (RURI fixes, etc.) rewritehostport("PSTN_IP:PSTN_PORT"); if (!is_method("BYE")) { t_on_reply("ROUTE_TO_PSTN_ONREPLY"); t_on_failure("ROUTE_TO_PSTN_FAILURE"); if (isflagset(FLAG_DEBUG)) xlog("L_INFO", "Relaying to PSTN - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci - On Reply, On Failure Set\n"); }; if (!t_relay()) { if (isflagset(FLAG_DEBUG)) xlog("L_INFO", "Relaying to PSTN - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci - Sending Reply Error\n"); sl_reply_error(); }; exit; } failure_route[ROUTE_TO_PSTN_FAILURE] { .SNIP. (SQL, etc for updating the CDR) if (t_check_status("408")) { rewriteuri("sip:ANC-408@ANS_IP:ANS_PORT"); append_branch(); t_relay(); exit; }; exit; }

1 0

[OpenSER-Users] remote IP address
by Victor Gamov 13 Aug '07

13 Aug '07

Hi All! Is it possible to know IP-address to which request will be forwarded ? I see that t_relay() actually know it but how get these knowledge in script? any pseudo-variable available for it? -- CU, Victor Gamov

4 9

[OpenSER-Users] Dispatcher round robin....
by Francisco Ivandic 13 Aug '07

13 Aug '07

Hi All, I'm using dispatcher succesfully, with round robin alg (option 4), and also use ds_mark_dst and ds_next_dst to select other proxy in case of failure. It balances great when all servers are active, but when there are inactive servers, round robin alg selects the next in the round, even if it's marked as inactive. This results in going through ds_next_dst, and overloads the first active server in the dispatcher file until completing the round. Then, other active servers are called, and when passing to a group of inactive servers, starts going through the first active (as much times as inactive servers there are), and again completing the round. Does anyone know a way to implement a round robin aware of inactive servers? Thanks in advance, Francisco Ivandic

1 0

[OpenSER-Users] subs_status is still active after unsubscription, is it right?
by KevinKinnan 13 Aug '07

13 Aug '07

Thank your Anca for subscribe tip!Notify request with terminated subscription status is got when unsubscribing. But the subs_status in watchers table is still active, is that right? Regards,Kevin _________________________________________________________________ 通过 Windows Live Spaces 与朋友轻松共享您的生活。 http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

2 1

[OpenSER-Users] trunk version 2592: NOTIFY problem
by KevinKinnan 13 Aug '07

13 Aug '07

SUBSCRIBE request can not save in watchers table properly. Both p_user and w_user are subscriber's name!? In this case, they are both 'alice'. The SUBSCRIBE request is shown as follows:----------------------------------------SUBSCRIBE sip:alice@148.88.232.101 SIP/2.0Call-ID: c14d16bf8f758655b09588b764c3f0c9(a)148.88.232.101CSeq: 20 SUBSCRIBEFrom: "alice" <sip:alice@148.88.232.101:5061>;tag=z9hG4bK41052461To: "bob" <sip:bob@148.88.232.101:5062>Via: SIP/2.0/TCP 148.88.232.101:5061;branch=z9hG4bK57563Max-Forwards: 70Expires: 360000Event: presenceRoute: <sip:148.88.233.209:5065;lr>Contact: "alice" <sip:alice@148.88.232.101:5061>Content-Length: 0----------------------------------------HELP!Regards,Kevin _________________________________________________________________ 通过 Live.com 查看资讯、娱乐信息和您关心的其他信息！ http://www.live.com/getstarted.aspx

2 1

[Serusers] preload_udomain: bad callid
by Steffen Waldmann 13 Aug '07

13 Aug '07

hi there, when i restart my ser i get the following error: 0(0) preload_udomain: ERROR: bad callid record in table location 0(0) preload_udomain: ERROR: for username 1000, contact sip:1000@192.168.2.117:30316;rinstance=80910b76433a8887 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad callid record in table location 0(0) preload_udomain: ERROR: for username 1001, contact sip:1001@192.168.2.220:5060;transport=udp 0(0) preload_udomain: ERROR: skipping... ser doesnt start up again... what could be the problem?

1 0

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