[Spce-user] General Comments and Questions

Andreas Granig agranig at sipwise.com
Wed May 23 17:50:12 EDT 2012


On 05/22/2012 06:58 PM, Klaus Darilion wrote:
> Congratulations to sip:provider, it is a great product and I really like
> the provided features and the way it is structured.

Thanks, that means a lot to us, coming from you!

> A) I would add usage of kamctl to the manual:
> 
> section 3.1.1.1
> 
> Add to last paragraph: For details about the Kamailio processes you can
> use the kamctl tool: 'OSER_FIFO=/var/run/kamailio/kamailio.lb.fifo
> kamctl ps'.

As Andrew pointed out in another thread, this is covered by using
"ngcp-kamctl <proxy|lb> <cmd>" and "ngcp-sercmd <proxy|lb> <cmd>".

> B) web password and sip password for subscribers
> 
> Probably brute-force hacking is mitigated by PIKE module, but the
> default password length of 6 characters is IMO to short and may be
> extended, eg. 12 characters. Especially SIP passwords need not be
> remembered as they are stored in the phone. It would also be cool if the
> web interface can help in choosing IDs (external IDs, web/sip usernames,
> passwords) e.g. by choosing subscriber usernames derived from account
> username and random passwords.

External IDs are purely optional, and I think as Andrew pointed out, you
can extend the minimum length of passwords. Auto-generating passwords is
something I wanted for quite some time, and the length for that could be
much longer than min pwd length. I'm priorizing this one, it's really
useful.

> C) IMO it is confusing that some parts require the E.164 number without
> leading + (e.g. subscriber's number), and some with leading + (e.g.:
> allowed_clis). For example when configuring "Peering Rules" I do not
> know what to put into the caller pattern, e.g. +43, or only 43 to use
> this route for calls to Austria. Of course reading the manual tells me
> to omit the +, but the web interface could be improved by using for
> example "string, caller pattern, e.g. 431 for calls to Vienna/Austria"
> instead of "string, pattern". One more: blocklists require a +4312345
> pattern whereas NCOS requires a 4312345 pattern (according to the manual).

Yeah, this is in our bugtracker for quite some time now. The thing here
is that the various interfaces try to be clever and normalize numbers,
however the plan is to let inputs go through 1:1 in the admin-panel
(because you as admin are supposed to know that you have to provide
everything in E164 format), whereas user input in the CSC goes through
the rewrite rules, so you have a consistent input format ("you get what
you dial").

> F) Is there a way to record conversations, e.g. for lawful intercept?

No. There are certain approaches we suppport in the PRO edition for
lawful interception, but none of that is provided in the CE.

> G) An included flash-player would be nice to playback the voicemail WAV
> files in the subscriber web interface.

Yeah, that is fairly easy, we'll add that for 2.6.

> H) For call-forwardings it would be nice if there would be a global
> "Destination Set" called "Voicebox/Voicemail/Sprachbox". IMO it is not
> intuitive for subscribers to first create a Destination Set to allow
> forwardings to the voicebox.

It used to be straight-forward in past released, but since 2.5 you can
define arbitrary call-forward scenarios including serial and time
definitions, so it got a bit complicated. Providing predefined
destination sets could work, let's see.

> J) Reloading web server config gives the warning:
> 
> apache2: Could not reliably determine the server's fully qualified
> domain name, using 127.0.0.1 for ServerName

Nobody complained so far, but we'll have a look how to avoid this :)

> K) The default voicemail template causes emails like:
> [Voicebox] New message 1 in voicebox 8d880382-00b8-411b-bb74-a91ec1e2819e
> "You have received a new message from sipmausi002 in voicebox
> 8d880382-00b8-411b-bb74-a91ec1e2819e on Tuesday, 22 May 2012 um 14:55:50."
> 
> IMO it would be nice if the default template shows the caller's and
> callee's phone numbers instead of the callee's subscriber-id and
> caller's SIP identity.

Looking into that as well. The idea here was that one subsciber could
have multiple alias numbers, but all should end up in the same voicebox.
We maybe can show the called number instead of the uuid.

Andreas

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