[asterisk-ss7] chan_ss7, sangoma A104 to local telecom partner, bridge to SIP, Hangup Cause 41

[Robert Verspuy]

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

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