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ty.roach@acecomm.com

2004-Nov-05 11:45 UTC

[Asterisk-Users] Questions from an Asterisk newbie

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 with two 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@172.20.23.201
To: sip:4444@172.20.23.201
Call-ID: ce30300-411dcd5-8f0953-2e323731@172.20.23.211
CSeq: 101 REGISTER
Contact: <sip:4444@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@172.20.23.201
To: sip:4444@172.20.23.201;tag=as106566ef
Call-ID: ce30300-411dcd5-8f0953-2e323731@172.20.23.211
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:4444@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@172.20.23.201' failed for
'172.20.23.211'
Scheduling destruction of call
'ce30300-411dcd5-8f0953-2e323731@172.20.23.211' in 15000 ms
Destroying call 'ce30300-411dcd5-8f0953-2e323731@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@172.20.23.201
To: sip:3005@172.20.23.201
Call-ID: 2ae30300-4302418-8f1c2b-2e323731@172.20.23.212
Date: Fri, 05 Nov 2004 18:38:54 GMT
CSeq: 101 REGISTER
Contact: <sip:3005@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@172.20.23.201
To: sip:3005@172.20.23.201;tag=as66d562fd
Call-ID: 2ae30300-4302418-8f1c2b-2e323731@172.20.23.212
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3005@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@172.20.23.201' failed for
'172.20.23.212'
Scheduling destruction of call
'2ae30300-4302418-8f1c2b-2e323731@172.20.23.212' in 15000 ms

Jean-Michel Hiver

2004-Nov-05 12:22 UTC

head link

[Asterisk-Users] Questions from an Asterisk newbie

Hi,
>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).
>I am a n00b too - Yet I had some limited success using this introduction:

http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html

Cheers,
Jean-Michel.

Paulo Adriano

2004-Nov-05 12:30 UTC

head link

[Asterisk-Users] Questions from an Asterisk newbie

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@wavelis.pt 
 

 
>>> ty.roach@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@172.20.23.201 
To: sip:4444@172.20.23.201 
Call-ID: ce30300-411dcd5-8f0953-2e323731@172.20.23.211 
CSeq: 101 REGISTER
Contact: <sip:4444@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@172.20.23.201 
To: sip:4444@172.20.23.201;tag=as106566ef 
Call-ID: ce30300-411dcd5-8f0953-2e323731@172.20.23.211 
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:4444@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@172.20.23.201' failed for
'172.20.23.211'
Scheduling destruction of call
'ce30300-411dcd5-8f0953-2e323731@172.20.23.211' in 15000 ms
Destroying call 'ce30300-411dcd5-8f0953-2e323731@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@172.20.23.201 
To: sip:3005@172.20.23.201 
Call-ID: 2ae30300-4302418-8f1c2b-2e323731@172.20.23.212 
Date: Fri, 05 Nov 2004 18:38:54 GMT
CSeq: 101 REGISTER
Contact: <sip:3005@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@172.20.23.201 
To: sip:3005@172.20.23.201;tag=as66d562fd 
Call-ID: 2ae30300-4302418-8f1c2b-2e323731@172.20.23.212 
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3005@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@172.20.23.201' failed for
'172.20.23.212'
Scheduling destruction of call
'2ae30300-4302418-8f1c2b-2e323731@172.20.23.212' in 15000 ms


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Ben Greear

2004-Nov-05 12:33 UTC

head link

[Asterisk-Users] Questions from an Asterisk newbie

ty.roach@acecomm.com wrote:> 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 with two 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 just did something similar.  I added these lines to /etc/asterisk/sip.conf:

; Grandstream
[1001]
type=friend
host=dynamic

; cisco phone
[1002]
type=friend
host=dynamic


Then I added these lines to /etc/asterisk/extensions.conf
exten => 1001,1,Dial(SIP/1001,200,tr)
exten => 1002,1,Dial(SIP/1002,200,tr)

My phones register as phone numbers 1001 and 1002.  There may be
a better way to do it, but with this config I was able to make
calls...

Ben


-- 
Ben Greear <greearb@candelatech.com>
Candela Technologies Inc  http://www.candelatech.com

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