<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content="text/html; charset=us-ascii" http-equiv=Content-Type>
<META name=GENERATOR content="MSHTML 8.00.6001.18852"></HEAD>
<BODY>
<DIV dir=ltr align=left><SPAN class=241261523-29122009><FONT color=#0000ff
size=2 face=Arial>I upgraded to wanpipe-3.5.8.7.tgz and Freeswitch version
1.0.5pre9 and the bug is still present. Would libpri possibly help?
I'm currently using the native wanpipe PRI stack and default openzap configs in
Freeswitch.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=241261523-29122009><FONT color=#0000ff
size=2 face=Arial></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=241261523-29122009><FONT color=#0000ff
size=2 face=Arial>Best Regards,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=241261523-29122009><FONT color=#0000ff
size=2 face=Arial>Jerry</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=241261523-29122009></SPAN> </DIV><BR>
<BLOCKQUOTE style="MARGIN-RIGHT: 0px">
<DIV dir=ltr lang=en-us class=OutlookMessageHeader align=left>
<HR tabIndex=-1>
<FONT size=2 face=Tahoma><B>From:</B> Anthony Minessale
[mailto:anthony.minessale@gmail.com] <BR><B>Sent:</B> Monday, December 28,
2009 3:31 PM<BR><B>To:</B>
freeswitch-users@lists.freeswitch.org<BR><B>Subject:</B> Re:
[Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed toVoice
Mail<BR></FONT><BR></DIV>
<DIV></DIV>you have to update the sangoma driver and probably FreeSWITCH for
good measure.<BR>Its a known bug in the sangoma driver that has been fixed it
the latest release.<BR><BR><BR><BR>
<DIV class=gmail_quote>On Mon, Dec 28, 2009 at 5:19 PM, Jerry Richards <SPAN
dir=ltr><<A
href="mailto:jerry.richards@teotech.com">jerry.richards@teotech.com</A>></SPAN>
wrote:<BR>
<BLOCKQUOTE
style="BORDER-LEFT: rgb(204,204,204) 1px solid; MARGIN: 0pt 0pt 0pt 0.8ex; PADDING-LEFT: 1ex"
class=gmail_quote>Hello All,<BR><BR>I posted a FS log into the Pastebin at
<A href="http://pastebin.freeswitch.org/11644"
target=_blank>http://pastebin.freeswitch.org/11644</A>.<BR><BR>I am still
having the problem where a PSTN-to-Internal call via a Sangoma<BR>A101D card
stops ringing the internal phone after about 10 seconds. It<BR>should
be ringing for 30 seconds and then go to Voice Mail (as
an<BR>Internal-to-Internal call does).<BR><BR>Best Regards,<BR><FONT
color=#888888>Jerry<BR></FONT>
<DIV>
<DIV></DIV>
<DIV class=h5><BR><BR>-----Original Message-----<BR>From: Jerry Richards
[mailto:<A
href="mailto:jerry.richards@teotech.com">jerry.richards@teotech.com</A>]<BR>Sent:
Tuesday, December 22, 2009 8:02 AM<BR>To: '<A
href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</A>'<BR>Subject:
PSTN-to-Internal Call Does Not Get Routed to Voice Mail<BR><BR><BR>I have a
Freeswitch PBX server with an installed Sangoma A101D card<BR>connected to a
PRI. Most everything works okay, however when I get an<BR>inbound call
from the PSTN, if the call is not answered within about 12<BR>seconds, the
call ends (so it doesn't go to voice mail). If I make a call<BR>from
one internal phone to another, then it will go to voice mail after
30<BR>seconds. How can I get the external call to route to voice mail
after 30<BR>seconds?<BR><BR>I put a new 11595 log into the pastebin.
Do you know any Freeswitch setting<BR>that might cause this?<BR><BR>If
this issue has been addressed before, what string should I use to
search<BR>for it, because I can't find
it.<BR><BR>Thanks,<BR>Jerry<BR><BR><BR>_______________________________________________<BR>FreeSWITCH-users
mailing list<BR><A
href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</A><BR><A
href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users"
target=_blank>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</A><BR>UNSUBSCRIBE:<A
href="http://lists.freeswitch.org/mailman/options/freeswitch-users"
target=_blank>http://lists.freeswitch.org/mailman/options/freeswitch-users</A><BR><A
href="http://www.freeswitch.org"
target=_blank>http://www.freeswitch.org</A><BR></DIV></DIV></BLOCKQUOTE></DIV><BR><BR
clear=all><BR>-- <BR>Anthony Minessale II<BR><BR>FreeSWITCH <A
href="http://www.freeswitch.org/">http://www.freeswitch.org/</A><BR>ClueCon <A
href="http://www.cluecon.com/">http://www.cluecon.com/</A><BR>Twitter: <A
href="http://twitter.com/FreeSWITCH_wire">http://twitter.com/FreeSWITCH_wire</A><BR><BR>AIM:
anthm<BR><A
href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</A><BR>GTALK/JABBER/<A
href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</A><BR>IRC:
<A href="http://irc.freenode.net">irc.freenode.net</A>
#freeswitch<BR><BR>FreeSWITCH Developer Conference<BR><A
href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</A><BR><A
href="http://iax:guest@conference.freeswitch.org/888">iax:guest@conference.freeswitch.org/888</A><BR><A
href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</A><BR>pstn:+19193869900<BR></BLOCKQUOTE></BODY></HTML>