There sbould be no relation in voice and sig as ss7 is a ccs type signalling. So voice circuits can go to any node. Vashkar. On Apr 22, 2012 6:19 PM, "her Garcia" <herlit11 at lycos.com> wrote: > Hi, thanks for your replies. I ?ll try your suggestions. The reason I have > differents adjpointcode and defaultdpc is that > my carrier has a config of non associated ss7, which means that signalling > goes exclusively to one node/switch and > the voice is carried to a different node/switch. > > I was wondering if I should set up two different channels in my > chan_dahdi.conf for each carrier?s switch? > > [channels] > #First channel signalling only > language=en > ... > sigchan=16 > adjpointcode = 8122-> signalling node > defaultdpc = 8122 -> signalling node > > #Second channel voice only > language=en > ... > adjpointcode = 8845 -> voice node > defaultdpc = 8845 -> voice node > channel= 1-15 > channel= 17-31 > > > Have you seen this carrier setting before? > > Thanks,regards > Hern?n > Apr 22, 2012 07:59:02 AM, asterisk-ss7 at lists.digium.com wrote: > > =========================================== > > > Try putting sane point code in the adpointcode as defaultdpc > On Apr 20, 2012 7:16 PM, "her Garcia" wrote: > Hi, everyone. I am working on asterisk+ss7. > When I try to make a call, the call connects but I have no audio or see no > progress in the debug. > > > -- Executing [111536972876 at incoming:2] > Dial("SIP/1153640000-00000005", "DAHDI/17") in new stack > -- Executing [111536972876 at incoming:2] > Dial("SIP/1153640000-00000005", "DAHDI/17") in new stack > host*CLI> -- Called DAHDI/17 > -- Called DAHDI/17 > > Nothing else. I believe it should also include the following: > > >> -- DAHDI/1-1 is proceeding passing it to SIP/600-08887770 --- I > don?t get this > >> -- DAHDI/1-1 is ringing > >> -- DAHDI/1-1 answered SIP/600-08887770 > > My linkset is up, my channels are ok. My carrier tells me that he doesn?t > see any calls reaching his node. > I believe it?s because the call doesn?t progress. This is my config > > > The carrier says that his ss7 is semi-associated. Divides signalling in > one node and voice trunks/circuits in > a second node. I only have the following to configure > > adjpointcode=8122 > defaultdpc=8845 > > I know defaultdpc is the remote end. Signalling is ok verified by my > carrier, so I think my adjpointcode is ok. > The thing is that I also get messages from a third node in my debug, > number "8923" saying the following: > > > WARNING[18934]: sig_ss7.c:392 ss7_find_cic_gripe: Linkset 1: SS7 RLC > requested unconfigured CIC/DPC 14/8923. > > I understand its about the circuits. I tried configuring that node as my > adjpointcode, but I can?t get through, it > maybe something on the Carrier side for this particular node(8923) > > I have been working this for a couple of weeks, any ideas? > Thanks, I apologize for this long post. > Hern?n > > -- > _____________________________________________________________________ > -- 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... URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20120422/3f74a9b7/attachment-0001.htm>