[Asterisk-Users] Asterisk to Asterisk using SIP?

[Blaine Cook]

I'm able to place calls; I'm doing that with the call file instead 
(placing it into the outgoing spool).. My problem is that I'm not sure 
what sip.conf should look like.

Part of my problem I think is that I'm very confused as to how to 
direct incoming calls to the correct context, and how to get 
authentication going properly.

Here are the relevant (I think!) bits:

asterisk incoming                                                     | 
        asterisk outgoing
                                                                         
             |
sip.conf:                                                               
       |         sip.conf:
                                                                         
             |
register => user:userpw at outgoing/s                  |        [user]
                                                                         
             |          type=user
[outgoing]                                                              
     |          context=from-sip
type=peer                                                               
   |         secret=userpw
context=to-sip                                                          
|
host=asteriskoutgoing                                           |
                                                                         
            |
extensions.conf:                                                      | 
        extensions.conf:
                                                                         
            |
[to-sip]                                                                
        |        [from-sip]
                                                                         
            |
extension => s,1,Playback(test)                          |        
extension => s,1,Answer
extension => s,2,Wait(3)                                       |        
extension => s,2,Record(temp:gsm)
extension => s,3,Playback(test2)                        |        
extension => s,3,Playback(test3)
extension => s,4,Hangup                                     |        
extension => s,4,Record(temp:gsm)
                                                                         
           |         extension => s,5,Hangup
                                                                         
           |
/var/spool/asterisk/outgoing/call:                        |
                                                                         
           |
Channel: SIP/outgoing                                         |
Context: to-sip                                                        |
Extension: s                                                            
|
Priority: 1                                                             
     |

What am I doing wrong here?!

Sorry for the formatting, I hope it will be more clear with the configs 
side-by-side. I've attached the two configs in case it's more 
convenient.

Thanks!

blaine.

-------------- next part --------------
sip.conf:

register => user:userpw at outgoing/s

[outgoing
type=peer
context=to-sip
host=asteriskoutgoing

--------------------
extensions.conf:

[to-sip]

extension => s,1,Playback(test)
extension => s,2,Wait(3)
extension => s,3,Playback(test2)
extension => s,4,Hangup

--------------------
/var/spool/asterisk/outgoing/call:

Channel: SIP/outgoing
Context: to-sip
Extension: s
Priority: 1
-------------- next part --------------
sip.conf:

[user]
type=user
context=from-sip
secret=userpw

--------------
extensions.conf:

[from-sip]

extension => s,1,Answer
extension => s,2,Record(temp:gsm)
extension => s,3,Playback(test3)
extension => s,4,Record(temp:gsm)
extension => s,5,Hangup
-------------- next part --------------

On Oct 27, 2004, at 6:32 PM, Mark Phillips wrote:

> Simple. Write the below line into your extensions.conf (modifying where
> required) and you're off.
>
> exten => 100,1,Dial(SIP/100 at other_machine's_address
>
> The corresponding extension on the target * box must exist as an entry
> in the sip.conf file. This is also where you define the context.
>
> Hope that helps and don't shoot me if I have it all wrong
>
>
> Mark
>
>
> On Wed, 2004-10-27 at 20:44, Blaine Cook wrote:
>> Hi all,
>>
>> I have a bit of a conundrum that I'm not quite sure how to resolve. 
>> I'm
>> trying to place calls from one Asterisk server to another, with no
>> other SIP devices present. The purpose is for load testing. I've tried
>> various configurations, and none of them seem to get me any closer.
>> Usually I get 404 Not Found or 403 Forbidden errors.
>>
>> I have two contexts: [from-sip] and [to-sip]  .. What I want to do is
>> use a call file like:
>>
>> Channel: SIP/outbound
>> Context: to-sip
>> Extension: s
>> Priority: 1
>>
>> on the machine placing the calls, that would connect to the second
>> machine, that would receive the call and place it in the [from-sip]
>> context.
>>
>> I'm sure the configs for this are very simple, but I'm at a loss as to
>> what they are. Extensive searches for examples have come up fruitless.
>>
>> thanks for any assistance!
>>
>> blaine.
>>
>> _______________________________________________
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> -- 
>
> Mark Phillips, G7LTT/KC2ENI
> Randolph, NJ
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