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*
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3892 DB Zeewolde
Tel.: 088-OMICRON (66 427 66)
http://www.exa-omicron.nl
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