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@incoming:2] Dial("SIP/1153640000-00000005",
"DAHDI/17") in new stack
-- Executing [111536972876@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.
Hernán
--
_____________________________________________________________________
-- 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
