[Asterisk-Users] Asterisk Between PBX and FXS

[Fernando Lujan]

Melcon Moraes wrote:
> What PBX is that? What do you see at CLI> when you call that port?
>
> Actually, this behavior exists in the PBX. I need to make asterisk works like
> a fxs that will receive the calls, and then route them in two diferents SIP.
> Using your example:
>
> I call to the extension 100 or extension 101. This will be routed to
> asterisk. Inside a dialplan, how could I know if the incomming call was to
> 100 ou 101?


This is produced when I receive a call. In the moment I receive the 
call, I need to know if it was dialed 101 or 100 in the first pbx, just 
before Asterisk.

    -- Starting simple switch on 'Zap/3-1'
    -- Executing Dial("Zap/3-1", "SIP/12345|10") in new stack
    -- SIP Seeding peer from astdb: '12345' at 12345 at 192.168.1.59:5061 
for 1800
    -- Called 12345
Mar 30 10:53:23 WARNING[5735]: channel.c:2698 
ast_channel_make_compatible: No path to translate from 
SIP/12345-8a30(256) to Zap/3-1(68)
Mar 30 10:53:23 WARNING[5735]: chan_sip.c:2530 sip_write: Asked to 
transmit frame type 4, while native formats is 256 (read/write = 256/256)
Mar 30 10:53:23 WARNING[5735]: chan_sip.c:2530 sip_write: Asked to 
transmit frame type 4, while native formats is 256 (read/write = 256/256)
    -- SIP/12345-8a30 is ringing
    -- SIP Seeding peer from astdb: '12345' at 12345 at 192.168.1.59:5061 
for 
1800                                                                             


Thanks in advance.

Fernando Lujan