[asterisk-bugs] [JIRA] (ASTERISK-23472) Wiki Documentation - Configuration/Channel Drivers/SIP

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23472?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=217225#comment-217225 ] 

Rusty Newton commented on ASTERISK-23472:
-----------------------------------------

Note for chan_sip examples. The sip.conf.sample file contains text discussing use of the "callbackextension" option:

{noformat}
;----------------------------------------- OUTBOUND SIP REGISTRATIONS  ------------------------
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
;       register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
;
;
;
; domain is either
;       - domain in DNS
;       - host name in DNS
;       - the name of a peer defined below or in realtime
; The domain is where you register your username, so your SIP uri you are registering to
; is username at domain
;
; If no extension is given, the 's' extension is used. The extension needs to
; be defined in extensions.conf to be able to accept calls from this SIP proxy
; (provider).
;
; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
; this is equivalent to having the following line in the general section:
;
;        register => username:secret at host/callbackextension
{noformat}

The {{callbackextension}} option's use is really not explained well here, as it requires definition of other options to work in certain scenarios.

Nor is it explained elsewhere in the sample file that I can find.
{noformat}
;callbackextension=123            ; Register with this server and require calls coming back to this extension
{noformat}

A proposed description for the wiki:

Having {{callbackextension}} defined for a peer will cause Asterisk to attempt registering to the {{host}} defined for that peer.
If the registrar requires authorization, then for the peer you must define {{defaultuser}} for the user name, and then {{remotesecret}} for the password. These credentials will be used in response to the registrar's authorization challenge.
When the remote server registered to calls this account, they should use what you defined in the {{callbackextension}} as the dialed extension.


> Wiki Documentation - Configuration/Channel Drivers/SIP
> ------------------------------------------------------
>
>                 Key: ASTERISK-23472
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23472
>             Project: Asterisk
>          Issue Type: Improvement
>      Security Level: None
>          Components: Documentation
>            Reporter: Rusty Newton
>            Severity: Critical
>
> https://wiki.asterisk.org/wiki/display/AST/SIP
> Need SIP channel driver content here.
> We have https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip , but may want to change the name of it, plus we need content for chan_sip.
> For chan_sip we should build an outline and then put it out there on the list or reviewboard for review to make sure we create something useful. We should also try to keep some consistency with the other channel driver documentation, as much as that makes sense.
> We'll also want at least a few example configuration scenarios for each channel driver. These should be more detailed than what is included in the configuration files. The configuration files should hold handy quick templates and not be an exhaustive guide.



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