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@xxxxxxxxxx> wrote:
On Wed, Jan 29, 2014 at 9:01 AM, Glen Millard <glenmillard@xxxxxxxxx> wrote:What is up on the wiki right now [1] is certainly more in the realm of
> 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.
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].
The CLI commands haven't been changed (although there are some new
> 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?
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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