[Asterisk-Users] Questions from an Asterisk newbie

[Paulo Adriano]

HI,
 
First you should start by editing  two different  .conf files: sip.conf
 and  extensions.conf    in /etc/asterisk
The best way is to let asterisk create the samples    command      make
samples   in   /sr/src/asterisk
Sample  .conf   files will be created in /etc/asterisk
 
A step by step info can be found in
http://www.voip-info.org/wiki-Asterisk+installation+tips    go to the
part of SIP configuration.   There is also some info about your Cisco
phones.
 
Good Luck
 
Paulo
 
Francisco Paulo Adriano
WaveLIS LDA
Mobile +351 91 870 87 98
Office + 351 21 989 83 34
Fax     +351 21 989 83 35
E-mail  :  pauloadriano at wavelis.pt 
 

 

>>> ty.roach at acecomm.com 05-11-2004 18:45:46 >>>

I have just installed asterisk in the hopes of operating a very simple
VoIP
demo.  The demo environment is as follows:

Asterisk 1.0.2 installed on a Fedora 2 Linux laptop.  The laptop is
connected to a hub along wittwo Cisco 7960 IP phones (SIP enabled). 
I've
manually configured the phones setting the IP address of the phones,
phone
names (extensions), the IP address of the SIP proxy (Asterisk
server?).

I have not made any modifications to any of the asterisk configuration
files.

I run asterisk ('asterisk -cv') from the command line just to see what
happens.  Essentially, I get messages from both SIP phones indicating
that
registration is failing (I guess not such as surprise since I haven't
configured anything).

For starters, I was hoping that some of the experts on this board
could
give me some tips on what I need to do to allow one phone to
successfully
call the other phone.  I did a similar thing several years ago using a
SIP
proxy server (from Dynamicsoft, albeit, with help from their support
group).

Any advise would be greatly appreciated.  Thanks and advance.

Ty

P.S.  I've included command line output from my asterisk console
below...


*CLI> sip debug
SIP Debugging Enabled
*CLI>
*CLI>
*CLI>
*CLI>

Sip read:
REGISTER sip:172.20.23.201 SIP/2.0
Via: SIP/2.0/UDP 172.20.23.211:5060
From: sip:4444 at 172.20.23.201 
To: sip:4444 at 172.20.23.201 
Call-ID: ce30300-411dcd5-8f0953-2e323731 at 172.20.23.211 
CSeq: 101 REGISTER
Contact: <sip:4444 at 172.20.23.211:5060>
Expires: 3600
Content-Length: 0


9 headers, 0 lines
Using latest request as basis request
Sending to 172.20.23.211 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.20.23.211:5060
From: sip:4444 at 172.20.23.201 
To: sip:4444 at 172.20.23.201;tag=as106566ef 
Call-ID: ce30300-411dcd5-8f0953-2e323731 at 172.20.23.211 
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:4444 at 172.20.23.201>
Content-Length: 0


to 172.20.23.211:5060
Nov  5 13:37:01 NOTICE[-159417424]: chan_sip.c:7571 handle_request:
Registration from 'sip:4444 at 172.20.23.201' failed for '172.20.23.211'
Scheduling destruction of call
'ce30300-411dcd5-8f0953-2e323731 at 172.20.23.211' in 15000 ms
Destroying call 'ce30300-411dcd5-8f0953-2e323731 at 172.20.23.211' 


Sip read:
REGISTER sip:172.20.23.201 SIP/2.0
Via: SIP/2.0/UDP 172.20.23.212:5060
From: sip:3005 at 172.20.23.201 
To: sip:3005 at 172.20.23.201 
Call-ID: 2ae30300-4302418-8f1c2b-2e323731 at 172.20.23.212 
Date: Fri, 05 Nov 2004 18:38:54 GMT
CSeq: 101 REGISTER
Contact: <sip:3005 at 172.20.23.212:5060>
Expires: 3600
Content-Length: 0


10 headers, 0 lines
Using latest request as basis request
Sending to 172.20.23.212 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.20.23.212:5060
From: sip:3005 at 172.20.23.201 
To: sip:3005 at 172.20.23.201;tag=as66d562fd 
Call-ID: 2ae30300-4302418-8f1c2b-2e323731 at 172.20.23.212 
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3005 at 172.20.23.201>
Content-Length: 0


to 172.20.23.212:5060
Nov  5 13:37:36 NOTICE[-159417424]: chan_sip.c:7571 handle_request:
Registration from 'sip:3005 at 172.20.23.201' failed for '172.20.23.212'
Scheduling destruction of call
'2ae30300-4302418-8f1c2b-2e323731 at 172.20.23.212' in 15000 ms


_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com 
http://lists.digium.com/mailman/listinfo/asterisk-users 
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041105/43decd57/attachment.htm