[Asterisk-Users] Re: sip to sip, not comunication

[Rich Adamson]

> log the call
> 
> -- Unregistered SIP '1234'
>     -- Registered SIP '1234' at 201.230.97.110 port 9906 expires 3600
>     -- Added extension '1234' priority 1 to sipregistrations
>     -- Saved useragent "xlite release 3010n stamp 19039" for peer 1234
>     -- Executing Dial("SIP/1234-cfb0", "SIP/123 at domain.com.pe|20|tr")
> in new stack
>     -- Called 123 at sorcier.com.pe
>     -- Unregistered SIP '12345'
>     -- Nobody picked up in 20000 ms
>   == Auto fallthrough, channel 'SIP/1234-cfb0' status is 'NOANSWER'

You will first need to make your sip phone "register" with asterisk.
That requires a username, secret and the IP address of your asterisk
box to be programmed into the sip phone, and obviously it must match
what you define in sip.conf.

Use 'sip show peers' to display those phones that have registered. Don't
bother trying to place a call until your phones show up on the display.

You might want to also visit voip-info.org as it has a lot of helpful
info for beginners. You should also find a list of pretty good references
at http://www.asterisk.org/support that will help.

If you installed asterisk from source code, you will also find a lot of
useful references in the /usr/src/asterisk/configs directory (sample
config files) as well as in /usr/src/asterisk/doc directory.

Rich