I have been using *
internally with SIP for about a month. So far, so good.
The next venture is
getting an outbound ZAP T1 to call out on an E&M MCI circuit. I am having
some issues whereas the call does not go anywhere. The signaling for the T1 is
good through zttool with no errors or alarms.
In some
documentation (on voip-info.org) there are some indications that I need to have
a file placed into /var/spool/asterisk/outbound for any call to go through. Is
this the case? If so, does anyone have pointers on doing this with
extensions.conf dynamically? Per call?
I have my dial plan
set as:
[outbound]
ignorepat =>
9
exten =>
_9.,1,Dial(${OUTBND1}/${EXTEN:1})
exten => _9, 2,
Congestion
exten => _9.,3,
Hangup
OUTBND1 is setup in
the globals section as ZAP/4 (405P Digium card - although the card shows up in
FC 1 as a 410P card). I have the other 3 T1s in the quad card setup for internal
calls and for some other telephony equipment for line conferencing. So far that
is all working well.
One thing I did
notice from /var/log/messages are messages that the sound card resource is being
disabled since it does not work well with non-full duplex sound cards. Do I need
to have an active, in use sound card for outbound dialing for the digit
tones?
Thanks to all in
advance.
No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.8 - Release Date: 5/10/2005
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
