[Asterisk-Users] problems with mediatrix 1204 FXO

Martin Pycko martinp at digium.com
Tue Sep 2 11:30:54 MST 2003


Try canreinvite=no

Martin

On Tue, 2 Sep 2003, Zac Sprackett wrote:

> I'm having a problem getting outbound trunking to work using asterisk
> and an external SIP FXO.
>
> 7 digit dialing produces the following output:
>
>  -- Executing Dial("SIP/mitel-fe17", "SIP/5925660 at mediatrix-1204") in new stack
>  -- Called 5925660 at mediatrix-1204
>  -- SIP/mediatrix-1204-645e answered SIP/mitel-fe17
>  -- Attempting native bridge of SIP/mitel-fe17 and SIP/mediatrix-1204-645e
>  -- Got SIP response 481 "Call-Leg Does Not Exist" back from 172.20.16.7
>  == Spawn extension (internal, 5925660, 1) exited non-zero on 'SIP/mitel-fe17'
>
> THe PSTN gateway is siezing the trunk and dialing the call.  The native
> bridging seems to be the point of failure.  The caller (another sip set)
> gets hung up on and the called pary hears dead air.
>
> A sip debug log of the scenario is available here:
>
> http://sprackett.com/asterisk.txt
>
> extensions.conf snippet:
>
> [trunks-local]
> ; local 7 digit dialing
> exten => _NXXXXXX,1,Dial(SIP/${EXTEN}@mediatrix-1204)
> exten => _NXXXXXX,2,Congestion
>
> [internal]
> include => trunks-local
>
>
> sip.conf snippet:
>
> [mediatrix-1204]
> type=peer
> host=172.20.16.7
> mask=255.255.255.255
> dtmfmode=inband
> context=default
> canreinvite=yes
> qualify=yes
>
> Thanks in advance for any adive you can give me.
> -z
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>




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