asterisk users - Aug 2009 - Help for Alcatel asterisk

Hello everybody

I have an asterisk with an integration of alcatel pbx, by sip trunk, all
calls are fine, but tha calls calls that originate from a analog line,
the recipient is not listening, and that if they hear the call originates,
the lines are E1 in alcatel pbx.

When a asteris user call to analog line the call is ok.


Everyone, has been that problem?

I change asterisk version 1.4.21 to 1.4.18 but the same problem.

I saw  the cli

[Aug 12 16:15:40] WARNING[2997]: chan_sip.c:3927 sip_indicate: Don't know
how to indicate condition 9
[Aug 12 16:15:40] WARNING[2997]: channel.c:2369 ast_indicate_data: Unable to
handle indication 9 for 'SIP/4001-0a16f5c0'

Anyone can help me..


Regards
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Hello Carlos,

I have the same problem when I try to do a flash hook with dahdi module.
Did you resolve your problem?


Thanks in advance,

Alex

2009/8/13 Carlos Rojas <crt.rojas at gmail.com>:
> Hello everybody
>
> I have an asterisk with an integration of alcatel pbx, by sip trunk, all
> calls are fine, but tha calls calls that originate from a analog line,
> the recipient is not listening, and that if they hear the call originates,
> the lines are E1 in alcatel pbx.
>
> When a asteris user call to analog line the call is ok.
>
>
> Everyone, has been that problem?
>
> I change asterisk version 1.4.21 to 1.4.18 but the same problem.
>
> I saw? the cli
>
> [Aug 12 16:15:40] WARNING[2997]: chan_sip.c:3927 sip_indicate: Don't
know
> how to indicate condition 9
> [Aug 12 16:15:40] WARNING[2997]: channel.c:2369 ast_indicate_data: Unable
to
> handle indication 9 for 'SIP/4001-0a16f5c0'
>
> Anyone can help me..
>
>
> Regards
>
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