[asterisk-users] Scaling voicemail

[Nathan Ward]

(I sent this previously, but my list membership was still processing so I don’t 
think it made its way out, apologies if it did!)

Hi all,

I’m building a voicemail system using Asterisk, with pjsip, and IMAP. I used to 
do a lot of Asterisk, but, catching up after a few years away.

We have existing voice infrastructure using Kamailio, registrations go there, 
not to Asterisk.

Functionality is mostly fine. Problem I’m having is scaling. I’m just testing 
stuff right now, and dumped in around 3500 mocked up voicemail users, and found 
that startup times and so on are quite long.

I’m wondering:
1) Is there a way to avoid defining an endpoint + AOR for each user, while 
retaining the ability to poll mailboxes, and sent NOTIFYs to users? I was 
hoping that I could at least define a generic AOR for our Kamailio server, 
however, there doesn’t seem to be a way to set the RURI and To: user on a 
per-endpoint basis for the MWI notifications, so the NOTIFYs go out with the 
RURI/To: set to a generic URI (which is something like sip:kamailio_ip:5060 
). I’d really like the ability to just say “send NOTIFYs 
to this SIP server with the mailbox (or better, some configurable value) as the 
user-part, but that doesn’t appear to be possible..?
2) Is there a way to spread mailbox polling out? All the mailboxes are checked 
at once, which means we get big load spikes, including at startup. It’d be 
great to have this spread out over time.
3) Does anyone have thoughts/ideas re. scaling an Asterisk voicemail-only 
server? Can it be done while retaining mailbox polling with the current 
infrastructure?

FWIW, I’m running 14.7.7, though that can easily be changed if there’s a better 
version to target for this.

--
Nathan Ward

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Re: [asterisk-users] pjsip: TOS not working any more

[Michael Maier]

On 08/09/2018 at 10:49 PM Michael Maier wrote:
> Hello!
> 
> I'm using TOS as shown below with pjsip 13.22.0-rc1 (same with 13.21.1).
> Unfortunately, the TOS isn't set in reality any more (it used to work
> some time ago).

Got the problem.

I'm using freepbx and for the configuration of TOS, I'm using the
individual config file pjsip.endpoint_custom_post.conf.

After disabling one of my trunks via GUI, this trunk couldn't be find
any more by asterisk in pjsip.endpoint_custom_post.conf and the complete
configuration provided there was ignored. I had to comment out the
deactivated trunk. Now it's working again.

But I'm wondering why I have to define TOS for each trunk individually
if it is already defined for the transport as defined below?


Thanks,
Michael


> 
> 
> Transport:        
> 
> ==
> 
> 
> Transport:  0.0.0.0-udp   udp  0    184  0.0.0.0:5061
> 
>  ParameterName  : ParameterValue
>  ==
>  allow_reload   : true
>  async_operations   : 1
>  bind   : 0.0.0.0:5061
>  ca_list_file   :
>  ca_list_path   :
>  cert_file  :
>  cipher :
>  cos    : 0
>  domain :
>  external_media_address :
>  external_signaling_address :
>  external_signaling_port    : 0
>  local_net  : 192.168.13.0/255.255.255.0
>  local_net  : 192.168.17.0/255.255.255.0
>  method : unspecified
>  password   :
>  priv_key_file  :
>  protocol   : udp
>  require_client_cert    : No
>  symmetric_transport    : false
>  tos    : 184
>  verify_client  : No
>  verify_server  : No
>  websocket_write_timeout    : 100
> 

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