[asterisk-app-dev] API/AMI Question

[Glen Millard]

I think I know what is happening.

Somehow the chan_sip module is not being loaded. If I go to the CLI, any of
the sip commands:

 *sip notify Send a notify packet to a SIP peer*
* sip prune realtime [peer|all] Prune cached Realtime users/peers*
*              sip qualify peer Send an OPTIONS packet to a peer*
*                    sip reload Reload SIP configuration*
*sip set debug {on|off|ip|peer} Enable/Disable SIP debugging*
*      sip set history {on|off} Enable/Disable SIP history*
*sip show {channels|subscriptio List active SIP channels or subscriptions*
*         sip show channelstats List statistics for active SIP channels*
*              sip show channel Show detailed SIP channel info*
*              sip show domains List our local SIP domains*
*              sip show history Show SIP dialog history*
*                sip show inuse List all inuse/limits*
*                  sip show mwi Show MWI subscriptions*
*              sip show objects List all SIP object allocations*
*                sip show peers List defined SIP peers*
*                 sip show peer Show details on specific SIP peer*
*             sip show registry List SIP registration status*
*                sip show sched Present a report on the status of the
scheduler queue*
*             sip show settings Show SIP global settings*
*                  sip show tcp List TCP Connections*
*                sip show users List defined SIP users*
*                 sip show user Show details on specific SIP user*


*                sip unregister Unregister (force expiration) a SIP peer
from the registry*
are not working.

So, at the risk of sounding like a fool, I want to build Asterisk 12
without using the new SIP stack (pjsip) and utilize *only* the chan_sip.

Not looking for hand-holding - if it in the Wiki, that would be excellent.

Can someone point me to that, please?

Thanks - Glen




On 29 January 2014 12:04, Matthew Jordan <mjordan at digium.com> wrote:

> On Wed, Jan 29, 2014 at 9:01 AM, Glen Millard <glenmillard at gmail.com>
> wrote:
> > Hello Dan et al;
> >
> > Okay - what I want to do:
> >
> > 1. discover what functionality that the Asterisk 12 REST/AMI has - what
> > would be the best way to do that please? I see the Wiki on
> Digium/Asterisk,
> > but maybe I am looking in the wrong spots. It does not seem to be clear.
> > Once I discover what it is capable of, then I can decide if I can use it
> > with what we need to do.
>
> What is up on the wiki right now [1] is certainly more in the realm of
> command reference information, so that people writing ARI applications
> have some documentation about what resources/operations are available
> to them. David has also written page [2] on getting started with ARI,
> which is a jumping off point for getting Asterisk configured
> correctly, putting a channel into Stasis, and doing basic control of
> the channel using ARI.
>
> We're working on some documentation that is more in line with 'how do
> I build things', but that's still a work in progress.
>
> In the meantime, you can look at the client libraries that have been
> written by various people, that are linked to off of that page. You
> may also want to look at some demo applications that were written by
> David in Python - these are available with the Python reference
> library up on GitHub [3].
>
> > 2. Asterisk 12 - I noticed that the CLI commands seem to be a whole new
> > 'language' almost. Is there a way to keep the old Asterisk 11 style and
> > before CLI commands?
>
> The CLI commands haven't been changed (although there are some new
> ones that have been added for PJSIP). What exactly are you referring
> to?
>
> [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ARI
> [2] https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI
> [3] https://github.com/asterisk/ari-py/tree/master/examples
>
>
>
> --
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
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