Robert Verspuy wrote: > When trying to forward an incoming SS7 call to a SIP channel. > > -- Executing [20xxxxxxx at from-ss7-kpnglasvezel:1] > Dial("SS7/KPN-Rt/1", "SIP/switch-1/2762") in new stack > -- Called switch-1/2762 > -- SIP/switch-1-0dd60c90 is making progress passing it to SS7/KPN-Rt/1 > Really destroying SIP dialog > '749a4bac1c8077d650593da1423cb29e at xxx.xxx.xxx.xxx' Method: INVITE > == Spawn extension (from-ss7-kpnglasvezel, xxxxxxxxx, 1) exited > non-zero on 'SS7/KPN-Rt/1' > -- SS7 hangup 'SS7/KPN-Rt/1' CIC=1 Cause=41 (state=7) > > When i create an audo dialout file (/var/spool/asterisk/outgoing) for > SIP/switch-1/2762 and then connecting it to the Echo application. > > By looking at the ISUP packets in wireshark I discovered the problem. For an incoming call I receive call I receive IAM (initiate call). When forwarding the call to a SIP channel, chan_ss7 sends back CPG (call progress). And then I received RSC (channel reset) back from the ss7 peer., which stops the whole call. When I changed the use_connect option of the linkset in the config to no, it will send back a ACM (address complete) after the IAM, then sens a CPG, and finnaly send a ANM (answer). My local telco probably needed the ACM. So I've got the SS7 working on both links! Ready for the network integration tests of my local telco. Regards, Robert -- *Exa-Omicron* Patroonsweg 10 3892 DB Zeewolde Tel.: 088-OMICRON (66 427 66) http://www.exa-omicron.nl
![]() |