[Spce-user] General Comments and Questions
Klaus Darilion
klaus.mailinglists at pernau.at
Tue May 22 12:58:37 EDT 2012
Hi!
Congratulations to sip:provider, it is a great product and I really like
the provided features and the way it is structured.
While playing with spce I wrote down my experiences. You might be
interested to e.g. improve documentation or the product.
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'.
section 3.1.1.2
Add to last paragraph: For details about the Kamailio processes you can
use the kamctl tool: 'OSER_FIFO=/var/run/kamailio/kamailio.proxy.fifo
kamctl ps'.
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.
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).
D) In input fields which expect a pattern/regexp it is unclear how the
string is interpreted, e.g. when defining Number Patterns for NCOS Level
and I want to block calls to Austria 09xx, what do I have to enter? The
mouse-over tool-tip tells me "POSIX regular expression", thus I would
need to enter ^439.* but the manual in section 5.1.2.2 tells me to use
439* which is not a POSIX expression but some proprietary pattern
matching. This confuses me in many input forms when defining
strings/patterns/regexps.
E) Is there a global option to 'always_use_rtpproxy' for ALL calls? I
guess forcing the proxy for all configured 'domains' should have the
same effect?
F) Is there a way to record conversations, e.g. for lawful intercept?
G) An included flash-player would be nice to playback the voicemail WAV
files in the subscriber web interface.
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.
I) The keep-alive OPTIONS requests contains the P-NGCP-Src-... Headers -
IMO they should not be sent to the subscribers:
OPTIONS sip:55.44.33.3:25488 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:55.44.32.175;r2=on;lr=on;ftag=32a293c3;ngcplb=yes>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=32a293c3;ngcplb=yes>
Via: SIP/2.0/UDP
83.136.32.175;branch=z9hG4bK859e.d70c6edae31c74d9691b07a7d56df889.0
Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0
From: sip:pinger at sipwise.local;tag=32a293c3
To: sip:55.44.33.3:25488
Call-ID: 8d170825-e2581a31-4d380f6 at 127.0.0.1
CSeq: 1 OPTIONS
Content-Length: 0
P-NGCP-Src-Ip: 127.0.0.1
P-NGCP-Src-Port: 5062
P-NGCP-Src-Proto: udp
P-NGCP-Src-Af: 4
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
It would be nice if this warning could be avoided (or did I missed some
configuration?).
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.
regards
Klaus
More information about the Spce-user
mailing list