[Asterisk-Users] I must be alone

[Greg Renouf]

It is funny that you mention this- when I tried to make a PSTN bridged
call through a E400 and a SIP phone this morning I had the exact same
problem.  I couldn't do a trace because I had to run out of the house
early.  Will trace next time it happens.
 
-GSR
 
-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Uriel
Carrasquilla
Sent: 08 April 2003 05:21
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] I must be alone
 
Hi everyone (Mark, Jim).  I am new to the list but thanks to both Mark
and Jim, I have being using "asterisk" since summer 2001.  I am just
updating my version that was a year old.  Yes, I know, I got busy with
other things like paying bills so I don't have to sleep with the dog
anymore.
 
Anyway,  one of the frustrations I have been dealing with (keep in mind
that my version of asterisk (it was called -ng), libpri, zaptel and
zapata are old) is that they leave the lines open (loopstart lines using
the kwelstart in asterisk and zapata) when I receive a call from the
PSTN and the asterisk PBX creates a bridge to connect to another line
(PSTN) going out.  No phones involved.  I have the old Zapata/Tormenta
ISA T1's (great job done by Jim) and I am using my unit as a PBX at
home.  I found that we were not ready for prime time yet so I have been
waiting.  The channel bank I use is the Atlas TA 750 with both FXO and
FXS cards.
I never had this problem when I had loopstart lines from the PSTN for
incoming calls and trunks (groundstart) lines for terminating calls in
the PSTN.  I just placed an order for those lines at home and I am going
to have to pay big time because they are putting a T1 into my house just
for this purpose (and charge me the installation but not the monthly, I
hope in time I can convert to a PRI-ISDN).
I also use the Wildcards (both USB and PCI versions) in another unit I
have in Colombia plus one in Jamaica.
 
If that was not enough, I do have an anoying problem when I am on a VoIP
call to another unit.  When a call comes in from the PSTN and it is
taken by, say my wife (the dog has not learned how to answer yet), I
find that consistently the VoIP goes simplex (i.e. the other side (with
asterisk and either wildcards or Tormenta) end up loosing reception.
 
am I alone?
 
Thank you guys.
(Jim, by the way, if you are out there, I still got you defined in iax).
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030408/938f1241/attachment.htm