Hi use libss7 ,dahdi,asterisk-1.6 and conform d-channel (1 or 16) on your E1 telco link. On Wed, May 26, 2010 at 3:47 PM, Basil Hweij <basil at audiotelecom.net> wrote: > Dear Sir, > > > > It's the first time we use SS7 signaling, we have sangoma A104 card and we > want to use it for our new SS7 connection with telco. > > Please I need help how I can apply libss7 parameters with one that telco > sent it as below: > > > > > > SS7/C7 PARAMETERS > > Link 1 System #1 > > Span # = 1st E1 > > Time Slot = 1 > > Originating Point Code = 1081 (MSC1) NI=3 > > Destination Point Code = 1056 (Audio telecom server) NI=3 > > SLC = 0 > > Link Type = F > > Note: Point codes above are in decimal format. > > > ******************************************************************************* > > ISUP SPECIFICATIONS ? ITU > > > ******************************************************************************* > > ISUP PARAMETERS: Enter your CIC mapping into the following table or > provide additional sheets with this information. > > CIC = 2-31 > > Span # = 1st E1 > > Timeslot = 2-31 > > > > CIC = 33-63 > > Span # = 2 nd E1 > > Timeslot = 1-31 > > > > CIC = 65-95 > > Span # = 3rd E1 > > Timeslot = 1-31 > > > > CIC = 97-127 > > Span # = 4th E1 > > Timeslot = 1-31 > > > ******************************************************************************** > > E1 configuration: > > FRAMING = No Multi frame structure (E1 only) > > LINE CODE = HDB3 > > SS7 = ITU SS7 > > SS7 link type = F link > > > > > > And our libss7 configuration: > > > > signaling=ss7 ;this is ss7 signaling > > ss7type=itu ;using the ITU variant > > ss7_called_nai=dynamic ;NAI for outgoing calls > > ss7_calling_nai=dynamic ;NAI for incoming calls > > ss7_internationalprefix=00 ;international prefix value for incoming calls > > ss7_nationalprefix=0 ;national prefix value for incoming calls > > ss7_subscriberprefix= ;subscriber prefix value for incoming calls > > ss7_unknownprefix= ;unknown prefix value for incoming calls > > ss7_explictacm=yes ;ACM is send as soon as call enters the dial plan...may not accepted yet though > > *linkset= ;arbitrary name for this set of channels*** > > *pointcode= ;the point code for this system...aka SPC*** > > *adjpointcode= ;the point code for the system that we are signaling to... aka APC*** > > *defaultdpc= ;the point code for the system that the CICs will be negotiated with...aka DPC*** > > *networkindicator=international ;NI value for MTP3*** > > *cicbeginswith= ;the starting value of the CICs*** > > *channel= ;the channels that are CICs*** > > *sigchan= ;the signaling channel*** > > > > > > > > > > -- > _____________________________________________________________________ > -- 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 > -- BIPIN RAGHUVANSHI OPERATION HEAD ASTERISK (DEVELOPMENT AND RESEARCH) WWW.EHORIZONS.IN 011-32323262 011-46334633 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100528/8e84b2b6/attachment-0001.htm
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