[Asterisk-Users] Simple SIP only Asterisk Configuration

[Pikoro]

It is registering with the sip provider.  When I do a CLI> sip show 
registry, it shows as registered.

I really think my problem is about contexts.  For some reason, I think I 
got the flow of how a call comes in and goes out works.

Is this approximately right?:

Incoming SIP Calls:
PSTN -> SIP Provider -> ASTERISK -> "incoming" context in [general] of 
sip.conf -> matching [incoming] in extensions.conf -> sent do wherever 
the dialpan says to go.

Outgoing SIP Calls:
Phone -> asterisk > outgoing context in [ext] of sip.conf -> matching 
[outgoing] context in extensions.con -> sent to do whatever the dialpan 
says to do.

If that's wrong, please let me know :D

Cheers

 
Carlos Alperin wrote:

> That shouldn't be complicate, but it looks like you 're not 
> registering with your provider. However, without the configuration 
> files, it is not much to do for help you.
>
>  
>
> Carlos Alperin
>
>  
>
> ------------------------------------------------------------------------
>
> *From:* asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of *Pikoro
> *Sent:* Wednesday, October 26, 2005 11:01 PM
> *To:* asterisk-users at lists.digium.com
> *Subject:* [Asterisk-Users] Simple SIP only Asterisk Configuration
>
>  
>
> Ok, I've been messing with asterisk for the last 3 weeks and I just 
> can't seem to figure this out.
>
> Everything I've read seems to state that asterisk will work "out of 
> the box" with only minor config changes when being used only for SIP 
> to SIP calls.
>
> The problem I am having is I cannot make outbound calls or receive 
> incoming calls over my sip-provider.  Asterisk registers properly, and 
> internal communications seem to work fine.
>
> I have, at one time or another, had either outgoing only, or incoming 
> only, but never both at once.  Unfortunately, I didn't know what I did 
> to make either of those work since I had made multiple adjustments and 
> had done a reload after each change.  For some reason, the incoming 
> calls only started working after restarting the computer so it could 
> have been any of 50 things I had changed.
>
> I am back to the sample config files.  Is there any kind of 
> walkthrough for a sip only setup?  I have seen SIP only touched on 
> briefly, with most of the documentation leaning torwards IAX 
> communications.
>
> Here is what I am trying to accomplish:
>
>    1. Asterisk server registers with our sip-provider for sip to pstn
>       local and international calls
>    2. Internal extensions 0, 200-210 can call eachother (of course)
>    3. Extensions 200-205 are in a Tech Support Queue
>    4. Extensions 206-210 are in a Customer Support Queue
>    5. Extension 0 is the operator or menu system (I guess this would
>       be s?)
>    6. All phones (for now) are x-ten soft phones
>    7. Each extension has voice mail
>    8. When a customer calls during office hours, they are presented
>       with a menu, press 1 for CS, press 2 for TS, or dial the
>       extension you wish to reach, etc...
>    9. Calls can be forwarded to other extensions
>   10. On-hold music is implemented
>
> I can handle doing everything on the list except for #1.
>
> If anyone can offer any suggestions, it would make me, and my boss, 
> very happy.
>
> Thanks in Advance
> Aaron
>
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