<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=iso-8859-1">
<META content="MSHTML 6.00.2900.3243" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV><FONT face=Arial size=2>hi:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>i'm a new of asterisk voip server, i compiling
without problem asterisk 1.4.18, and other software and component.</FONT></DIV>
<DIV><FONT face=Arial size=2>i create two extension 20000 and 20100... and 30000
voicemailMain</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>but i can't call any extension this is the
logs</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>/var/logs/asterisk/messages</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>[Apr 7 13:25:19] WARNING[24402] app_dial.c:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)<BR>[Apr 7 13:26:27] WARNING[24407] app_dial.c: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)<BR>[Apr 7
13:26:51] NOTICE[24408] cdr.c: CDR simple logging enabled.<BR>[Apr 7
13:26:51] NOTICE[24408] loader.c: 144 modules will be loaded.<BR>[Apr 7
13:26:51] WARNING[24408] res_smdi.c: No SMDI interfaces are available to listen
on, not starting SMDI listener.<BR>[Apr 7 13:26:51] NOTICE[24408]
pbx_ael.c: Starting AEL load process.<BR>[Apr 7 13:26:51] NOTICE[24408]
pbx_ael.c: AEL load process: calculated config file name
'/etc/asterisk/extensions.ael'.<BR>[Apr 7 13:26:51] NOTICE[24408]
pbx_ael.c: AEL load process: parsed config file name
'/etc/asterisk/extensions.ael'.<BR>[Apr 7 13:26:51] NOTICE[24408]
pbx_ael.c: AEL load process: checked config file name
'/etc/asterisk/extensions.ael'.<BR>[Apr 7 13:26:51] NOTICE[24408]
pbx_ael.c: AEL load process: compiled config file name
'/etc/asterisk/extensions.ael'.<BR>[Apr 7 13:26:51] NOTICE[24408]
pbx_ael.c: AEL load process: merged config file name
'/etc/asterisk/extensions.ael'.<BR>[Apr 7 13:26:51] NOTICE[24408]
pbx_ael.c: AEL load process: verified config file name
'/etc/asterisk/extensions.ael'.<BR>[Apr 7 13:26:51] WARNING[24408]
chan_iax2.c: Unable to open IAX timing interface: No such file or
directory<BR>[Apr 7 13:27:02] WARNING[24439] file.c: File vm-login does
not exist in any format<BR>[Apr 7 13:27:02] WARNING[24439] file.c: Unable
to open vm-login (format 0x2 (gsm)): No such file or directory<BR>[Apr 7
13:27:02] WARNING[24439] app_voicemail.c: Couldn't stream login
file<BR>[Apr 7 13:27:48] NOTICE[24419] chan_sip.c: Call from '20000' to
extension '500' rejected because extension not found.<BR>[Apr 7 13:27:59]
WARNING[24440] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No
route to destination)</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>extensions.conf</FONT></DIV>
<DIV><FONT face=Arial
size=2>[globals]<BR>CONSOLE=Console/dsp ; Console
interface for demo</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial
size=2><BR>MACHINE1=SIP/20000<BR>MACHINE2=SIP/20100</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>;My
Extensions<BR>[ejemplo]<BR>;Yanier<BR>exten=>20000,1,Dial(${MACHINE1},30,Tm)<BR>exten=>20000,2,Hangup<BR>exten=>20000,102,Voicemail(20000)<BR>exten=>20000,103,Hangup</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial
size=2>;Pedro<BR>exten=>20100,1,Dial(${MACHINE2},30,Tm)<BR>exten=>20100,2,Hangup<BR>exten=>20100,102,Voicemail(20100)<BR>exten=>20100,103,Hangup</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial
size=2>;Other<BR>exten=>30000,1,VoicemailMain</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>SIP.conf</FONT></DIV>
<DIV><FONT face=Arial size=2>;Test
conf<BR>[20000]<BR>type=friend<BR>secret=a20000b<BR>qualify=yes<BR>nat=no<BR>canreinvite=no<BR>context=ejemplo<BR><A
href="mailto:mailbox=20000@ejemplobuzon">mailbox=20000@ejemplobuzon</A><BR>callerid=Yanier<BR>disallow=all<BR>allow=gsm</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial
size=2>[20100]<BR>type=friend<BR>secret=a20100b<BR>qualify=yes<BR>nat=no<BR>canreinvite=no<BR>context=ejemplo<BR><A
href="mailto:mailbox=20100@ejemplobuzon">mailbox=20100@ejemplobuzon</A><BR>callerid=Pedro<BR>disallow=all<BR>allow=gsm</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>voicemail.conf</FONT></DIV>
<DIV><FONT face=Arial
size=2>[primerbuzon]<BR>20000=>1234,Yanier,yanier@micorreo.com</FONT></DIV>
<DIV><FONT face=Arial size=2>20100=>4321,Juan,juan@micorreo.com</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>someone can helpme</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>PD: Sorry for my bad english.</FONT></DIV>
<DIV><FONT face=Arial size=2>PD2: someone can explain how to install
correct asterisk with some configuration examples(only for pc
lan).</FONT></DIV><br>Obe Provincial Ciego de Avila<br>
Ave de los Deportes, esq. Circunvalación Norte<br>
Telef: 200708<br>
</BODY></HTML>