[asterisk-ss7] Call conected but not ringing

her Garcia herlit11 at lycos.com
Fri Apr 20 08:15:53 CDT 2012

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


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.

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