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