Hello chan_dahdi and libss7 cant restrict the callerid - IMHO (asterisk 1.6.1.10 - dahdi 2.2.0, libss7 1.0.2, ss7 mode: ITU) the config file (chan_dahdi.conf) contains usecallingpres=yes - debugging the value on chan_dahdi line 14531 shows the config file is correctly read. however, chan_dahdi.c, line 9131 - tmp->use_callingpres = conf->chan.use_callingpres - does not retrieve the value from the config file and is awlays 0. I cant seem to find the relation between both - but hardcoding this to 1, and then setting screening indicator correctly allow the function to work. how do I report a bug ? J. On Thu, Oct 8, 2009 at 9:20 PM, Jean C?rien <cerien.jean at gmail.com> wrote: > > Hello > > Still searching on this one... I've tried to have chan_ss7 on the calling > box, and lib_ss7 on the other. > > well, chan_ss7 sets correctly the presentation & screening bits. When > libss7 / debug mode dumps the IAM message, on the called box, it says > 'Presentation : 1, screening 3', instead of Pres:0 Screen: 1 when using > lib_ss7 on the calling box. > > I've tested to switch to ansi mode, that does not affect it. It really > looks like there is a bug here ! > > > Help ! > > J. > > On Thu, Oct 8, 2009 at 7:08 AM, Jean C?rien <cerien.jean at gmail.com>wrote: > >> thanks for the help, I appreciate this, >> >> 1. the otherside is an asterisk box, linked directly by a crossover E1 - >> both boxes are running the same libss7 version. dumping the SS7 at the side >> initiating the call show no changes on the presentation/screening >> indicators. >> >> 2. no too sure what you mean - the lines simply dump the * variables >> output of the script is: >> ==> CallerId(all) : "2115" <2115> >> >> ==> CallerId(pres) : allowed_not_screened >> ==> CallerId(all) : "2115" <2115> >> ==> CallerId(pres) : prohib >> which tend to show that the variable change indeed is effective. >> >> My gut feeling is that SS7/ITU number presentation may not work correctly. >> J. >> >> >> On Wed, Oct 7, 2009 at 5:49 PM, Mesbahuddin Malik <mesbah.malik at gmail.com >> > wrote: >> >>> Two Things >>> >>> 1. Does your Otherside (TP -Termination Point ) accept Caller ID >>> restriction ? >>> >>> 2.At your Sipphone Context review Line 2 and 3 >>> >>> >>> Rgds >>> Mesbah >>> >>> >>> On 10/8/09, Jean C?rien <cerien.jean at gmail.com> wrote: >>>> >>>> >>>> Here it is ! >>>> 88 is the prefix i use to select the SS7 trunk when dialing. >>>> >>>> J. >>>> >>>> [general] >>>> static = yes >>>> writeprotect = no >>>> clearglobalvars = yes >>>> [globals] >>>> FEATURES = >>>> DIALOPTIONS = >>>> RINGTIME = 20 >>>> FOLLOWMEOPTIONS = >>>> >>>> [sipphones] >>>> exten => _88.,1,Verbose(1,************************ DAHDI ${EXTEN:1} ) >>>> exten => _88.,n,Verbose(1,==> CallerId(all) : ${CALLERID(all)}) >>>> exten => _88.,n,Verbose(1,==> CallerId(pres) : ${CALLERID(pres)}) >>>> exten => _88.,n,Set(CALLERPRES()=prohib) >>>> exten => _88.,n,Verbose(1,==> CallerId(all) : ${CALLERID(all)}) >>>> exten => _88.,n,Verbose(1,==> CallerId(pres) : ${CALLERID(pres)}) >>>> exten => _88.,n,Dial(dahdi/r1/${EXTEN:2}) >>>> >>>> >>>> >>>> >>>> On Wed, Oct 7, 2009 at 4:39 PM, Mesbahuddin Malik < >>>> mesbah.malik at gmail.com> wrote: >>>> >>>>> Hi, >>>>> Post ur extension.conf here >>>>> >>>>> Rgds >>>>> Mesbah >>>>> >>>>> >>>>> On 10/8/09, Jean C?rien <cerien.jean at gmail.com> wrote: >>>>> >>>>>> >>>>>> Hello >>>>>> >>>>>> I am struggling to restrict the presentation of the callerid with >>>>>> libss7. It is always presented, even when I set CALLERPRES()=prohib >>>>>> >>>>>> My setup: >>>>>> Sip phone1 -> A* #1 -> dahdi/libss7 -> e1 -> dahdi/libss7 -> A* #2 -> >>>>>> Sip Phone 2 >>>>>> A* : 1.6.1.6 >>>>>> Dahdi : 2.2.0 >>>>>> libss7 1.0.2 >>>>>> OS: Centos5 x64 >>>>>> >>>>>> I've tried all possible values for CallerPres - still it shows up at >>>>>> the other end. >>>>>> >>>>>> on the near side, with verbose, I see that the value is correctly set: >>>>>> -- Executing [88123 at sipphones:2] Verbose("SIP/2115-0823cc98", >>>>>> "1,==> CallerId(all) : "2115" <2115>") in new stack >>>>>> ==> CallerId(all) : "2115" <2115> >>>>>> -- Executing [88123 at sipphones:3] Verbose("SIP/2115-0823cc98", >>>>>> "1,==> CallerId(pres) : allowed_not_screened") in new stack >>>>>> ==> CallerId(pres) : allowed_not_screened >>>>>> -- Executing [88123 at sipphones:4] Set("SIP/2115-0823cc98", >>>>>> "CALLERPRES()=prohib") in new stack >>>>>> -- Executing [88123 at sipphones:5] Verbose("SIP/2115-0823cc98", >>>>>> "1,==> CallerId(all) : "2115" <2115>") in new stack >>>>>> ==> CallerId(all) : "2115" <2115> >>>>>> -- Executing [88123 at sipphones:6] Verbose("SIP/2115-0823cc98", >>>>>> "1,==> CallerId(pres) : prohib") in new stack >>>>>> ==> CallerId(pres) : prohib >>>>>> >>>>>> Far side SS7 debug:(this part is identical in the near side as well) >>>>>> Calling Party Number: >>>>>> Nature of address: 3 >>>>>> NI: 0 >>>>>> Numbering plan: 1 >>>>>> Presentation: 0 >>>>>> Screening: 1 >>>>>> Address signals: 2115 >>>>>> [ 0a 04 03 11 12 51 ] >>>>>> >>>>>> Presentation & screening NEVER change when I change the pres value ! >>>>>> (I obviously reload the dial plan between each test) >>>>>> chan_dahdi.conf has >>>>>> usecallingpres=yes >>>>>> usecallerid=yes >>>>>> >>>>>> Two things worth noting: >>>>>> - Set(CALLERID(num)=012345 works >>>>>> - the presentation blocking works with chan_ss7 >>>>>> >>>>>> >>>>>> Would you have an idea as to why the number can not be hidden ? >>>>>> >>>>>> Many thanks, >>>>>> >>>>>> J. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >>>>>> >>>>>> asterisk-ss7 mailing list >>>>>> To UNSUBSCRIBE or update options visit: >>>>>> http://lists.digium.com/mailman/listinfo/asterisk-ss7 >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >>>>> >>>>> asterisk-ss7 mailing list >>>>> To UNSUBSCRIBE or update options visit: >>>>> http://lists.digium.com/mailman/listinfo/asterisk-ss7 >>>>> >>>> >>>> >>>> _______________________________________________ >>>> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >>>> >>>> asterisk-ss7 mailing list >>>> To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-ss7 >>>> >>> >>> >>> _______________________________________________ >>> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >>> >>> asterisk-ss7 mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-ss7 >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... 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