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<DIV>HI,</DIV>
<DIV> </DIV>
<DIV>First you should start by editing two different .conf
files: sip.conf and extensions.conf in
/etc/asterisk</DIV>
<DIV>The best way is to let asterisk create the samples
command make samples in
/sr/src/asterisk</DIV>
<DIV>Sample .conf files will be created in /etc/asterisk</DIV>
<DIV> </DIV>
<DIV>A step by step info can be found in <A
href="http://www.voip-info.org/wiki-Asterisk+installation+tips">http://www.voip-info.org/wiki-Asterisk+installation+tips</A>
go to the part of SIP configuration. There is also some info
about your Cisco phones.</DIV>
<DIV> </DIV>
<DIV>Good Luck</DIV>
<DIV> </DIV>
<DIV>Paulo</DIV>
<DIV> </DIV>
<DIV>Francisco Paulo Adriano<BR>WaveLIS LDA<BR>Mobile +351 91 870 87
98<BR>Office + 351 21 989 83 34<BR>Fax +351 21 989 83
35<BR>E-mail : <A
href="mailto:pauloadriano@wavelis.pt">pauloadriano@wavelis.pt</A></DIV>
<DIV> </DIV>
<DIV><BR> </DIV>
<DIV><BR>>>> ty.roach@acecomm.com 05-11-2004 18:45:46
>>><BR></DIV>
<DIV>I have just installed asterisk in the hopes of operating a very simple
VoIP<BR>demo. The demo environment is as follows:<BR><BR>Asterisk 1.0.2
installed on a Fedora 2 Linux laptop. The laptop is<BR>connected to a hub
along wittwo Cisco 7960 IP phones (SIP enabled). I've<BR>manually
configured the phones setting the IP address of the phones, phone<BR>names
(extensions), the IP address of the SIP proxy (Asterisk server?).<BR><BR>I have
not made any modifications to any of the asterisk
configuration<BR>files.<BR><BR>I run asterisk ('asterisk -cv') from the command
line just to see what<BR>happens. Essentially, I get messages from both
SIP phones indicating that<BR>registration is failing (I guess not such as
surprise since I haven't<BR>configured anything).<BR><BR>For starters, I was
hoping that some of the experts on this board could<BR>give me some tips on what
I need to do to allow one phone to successfully<BR>call the other phone. I
did a similar thing several years ago using a SIP<BR>proxy server (from
Dynamicsoft, albeit, with help from their support<BR>group).<BR><BR>Any advise
would be greatly appreciated. Thanks and
advance.<BR><BR>Ty<BR><BR>P.S. I've included command line output from my
asterisk console below...<BR><BR><BR>*CLI> sip debug<BR>SIP Debugging
Enabled<BR>*CLI><BR>*CLI><BR>*CLI><BR>*CLI><BR><BR>Sip
read:<BR>REGISTER sip:172.20.23.201 SIP/2.0<BR>Via: SIP/2.0/UDP
172.20.23.211:5060<BR>From: sip:4444@172.20.23.201<BR>To:
sip:4444@172.20.23.201<BR>Call-ID:
ce30300-411dcd5-8f0953-2e323731@172.20.23.211<BR>CSeq: 101 REGISTER<BR>Contact:
<sip:4444@172.20.23.211:5060><BR>Expires: 3600<BR>Content-Length:
0<BR><BR><BR>9 headers, 0 lines<BR>Using latest request as basis
request<BR>Sending to 172.20.23.211 : 5060 (non-NAT)<BR>Transmitting (no
NAT):<BR>SIP/2.0 403 Forbidden<BR>Via: SIP/2.0/UDP 172.20.23.211:5060<BR>From:
sip:4444@172.20.23.201<BR>To: sip:4444@172.20.23.201;tag=as106566ef<BR>Call-ID:
ce30300-411dcd5-8f0953-2e323731@172.20.23.211<BR>CSeq: 101
REGISTER<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:4444@172.20.23.201><BR>Content-Length:
0<BR><BR><BR>to 172.20.23.211:5060<BR>Nov 5 13:37:01 NOTICE[-159417424]:
chan_sip.c:7571 handle_request:<BR>Registration from 'sip:4444@172.20.23.201'
failed for '172.20.23.211'<BR>Scheduling destruction of
call<BR>'ce30300-411dcd5-8f0953-2e323731@172.20.23.211' in 15000
ms<BR>Destroying call
'ce30300-411dcd5-8f0953-2e323731@172.20.23.211'<BR><BR><BR>Sip read:<BR>REGISTER
sip:172.20.23.201 SIP/2.0<BR>Via: SIP/2.0/UDP 172.20.23.212:5060<BR>From:
sip:3005@172.20.23.201<BR>To: sip:3005@172.20.23.201<BR>Call-ID:
2ae30300-4302418-8f1c2b-2e323731@172.20.23.212<BR>Date: Fri, 05 Nov 2004
18:38:54 GMT<BR>CSeq: 101 REGISTER<BR>Contact:
<sip:3005@172.20.23.212:5060><BR>Expires: 3600<BR>Content-Length:
0<BR><BR><BR>10 headers, 0 lines<BR>Using latest request as basis
request<BR>Sending to 172.20.23.212 : 5060 (non-NAT)<BR>Transmitting (no
NAT):<BR>SIP/2.0 403 Forbidden<BR>Via: SIP/2.0/UDP 172.20.23.212:5060<BR>From:
sip:3005@172.20.23.201<BR>To: sip:3005@172.20.23.201;tag=as66d562fd<BR>Call-ID:
2ae30300-4302418-8f1c2b-2e323731@172.20.23.212<BR>CSeq: 101
REGISTER<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:3005@172.20.23.201><BR>Content-Length:
0<BR><BR><BR>to 172.20.23.212:5060<BR>Nov 5 13:37:36 NOTICE[-159417424]:
chan_sip.c:7571 handle_request:<BR>Registration from 'sip:3005@172.20.23.201'
failed for '172.20.23.212'<BR>Scheduling destruction of
call<BR>'2ae30300-4302418-8f1c2b-2e323731@172.20.23.212' in 15000
ms<BR><BR><BR>_______________________________________________<BR>Asterisk-Users
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