asterisk users - Jul 2004 - Need configuration sample for VoIP(SIP) -> PSTN Gateway

Hello,

 

I'm very new with * and I would really appreciate some help to implement a
SIP to PSTN Gateway.

My current scenario includes an * box with a TE405P board. I have a 1.5Mb
connection to the outside world (using a router with firewall capabilities) and
channel banks that allow me to connect the T1s coming out of the TE405 board to
the PSTN network (carrier).

I need to configure * to accept calls coming over IP (SIP) and terminate them
thru the Zaptel interface on the PSTN network.

Also need which parameters the SIP peer needs to know to connect to my system,
other than my IP and the port (ie: compression, codecs, etc.) and where/how to
configure those settings on my * box.

I have some understanding on how the configurations files work in *, what I
really need is some sample implementation that works for what I described, to
use them as a starting point for configuring my system.

Any help will be really appreciated.

Thanks in advance,

 

Alejandro.

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Can anyone explain the following IE 23 error:

Jul 17 09:27:27 VERBOSE[-1273529424]:     -- Zap/1-1 answered Zap/26-1
Jul 17 09:27:27 VERBOSE[-1273529424]:     -- Attempting native bridge of
Zap/26-1 and Zap/1-1
Jul 17 09:27:27 DEBUG[-1273529424]: master: 26, slave: 1, nothingok: 0
Jul 17 09:27:27 DEBUG[-1273529424]: Stoping tones on 26/0 talking to 1/0
Jul 17 09:27:27 DEBUG[-1273529424]: Stoping tones on 1/0 talking to 26/0
Jul 17 09:27:27 DEBUG[-1273529424]: Making 1 slave to master 26 at 0
Jul 17 09:27:27 DEBUG[-1273529424]: Added 19 to conference 9/26
Jul 17 09:27:27 DEBUG[-1273529424]: Added 43 to conference 9/1
Jul 17 09:27:27 DEBUG[-1273529424]: Updated conferencing on 26, with 0
conference users
Jul 17 09:27:27 DEBUG[-1273529424]: Updated conferencing on 1, with 0 conference
users
Jul 17 09:27:30 VERBOSE[-1210590288]: !! Unknown IE 23 (cs0, Unknown Information
Element)

I have a legacy IVR system we need to connect to Asterisk.  Our configuration
has two T100P cards. One connected to legacy IVR the other to the incoming CO
PRI.  A call is placed coming in via SIP extension and is answered by the IVR
system.  The IVR system then attempts to call out the CO PRI.  The call rings
the telephone but conversation paths is not complete when answering the call. 
The IE 23 error appears just before telephone rings.

I have also duplicate this by sending the call out an x100P.  The phone number
is dialed, the telephone rings, answering the telephone the connection stays up
until I hangup but the conversation path is not completed.

zaptel.conf
span=1,1,0,esf,b8zs,yellow
bchan=1-23
dchan=24

span=2,0,0,esf,b8zs,yellow
bchan=25-47
dchan=48

zapata.conf

switchtype=national
signalling=pri_cpe
group=2
channel=>1-23

switchtype=national
signalling=pri_net
group=3
channel=>25-47

;x100p
group=1
signalling=fxs_ks
channel=49

Lee
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